Script started on Fri 18 Feb 2005 15:58:39 GMT
]0;root@telth9:/tmp[root@telth9 tmp]# asterisk -r

Asterisk CVS-HEAD-10/20/04-18:58:37, Copyright (C) 1999-2004 Digium.

Written by Mark Spencer <markster@digium.com>

=========================================================================

Connected to Asterisk CVS-HEAD-10/20/04-18:58:37 currently running on telth9 (pid = 5847)
telth9*CLI>   Verbosity is atleast 2147483647

[Ktelth9*CLI>  
Sip read: 
INVITE sip:02070433288@incomingnum1.acmechat.org SIP/2.0 Via: SIP/2.0/UDP 213.166.5.129:5060;branch=z9hG4bK017f020b From: "07904306012" <sip:07904306012@213.166.5.129>;tag=as2bce6bbd To: <sip:02070433288@incomingnum1.acmechat.org> Contact: <sip:07904306012@213.166.5.129> Call-ID: 59d46e194f91c1c46a6857427b2d5c69@213.166.5.129 CSeq: 102 INVITE User-Agent: MSS VoIP Gateway Date: Fri, 18 Feb 2005 15:51:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 265  v=0 o=root 30707 30707 IN IP4 213.166.5.129 s=session c=IN IP4 213.166.5.129 t=0 0 m=audio 21698 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - 
 12 headers, 12 lines
 Using latest request as basis request
 Sending to 213.166.5.129 : 5060 (NAT)
 Found RTP audio format 8
 Found RTP audio format 0
 Found RTP audio format 3
 Found RTP audio format 101
 Peer audio RTP is at port 213.166.5.129:21698
 Found description format PCMA
 Found description format PCMU
 Found description format GSM
 Found description format telephone-event
 Capabilities: us - 0x50e(GSM|ULAW|ALAW|G729A|ILBC), peer - audio=0xe(GSM|ULAW|ALAW)/video=0x0(EMPTY), combined - 0xe(GSM|ULAW|ALAW)
 Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723)
 Found no matching peer or user for '213.166.5.129:5060'
 Looking for 02070433288 in prepaid
 list_route: hop: <sip:07904306012@213.166.5.129>
 Transmitting (NAT):
SIP/2.0 100 Trying Via: SIP/2.0/UDP 213.166.5.129:5060;branch=z9hG4bK017f020b;received=213.166.5.129;rport=5060 From: "07904306012" <sip:07904306012@213.166.5.129>;tag=as2bce6bbd To: <sip:02070433288@incomingnum1.acmechat.org>;tag=as6ca4c6df Call-ID: 59d46e194f91c1c46a6857427b2d5c69@213.166.5.129 CSeq: 102 INVITE User-Agent: acme VoIP Gateway Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:02070433288@217.14.132.185> Content-Length: 0  
 to 213.166.5.129:5060
 
[Ktelth9*CLI> 
    -- Executing [1;36;40mAGI[0;37;40m("[1;35;40mSIP/213.166.5.129-47a6a2a8[0;37;40m", "[1;35;40mlookup_dialplan.pl|02070433288[0;37;40m") in new stack
 
[Ktelth9*CLI> 
    -- Launched AGI Script /var/lib/asterisk/agi-bin/lookup_dialplan.pl
 
[Ktelth9*CLI> 
    -- AGI Script lookup_dialplan.pl completed, returning 0
 
[Ktelth9*CLI> 
    -- Executing [1;36;40mDial[0;37;40m("[1;35;40mSIP/213.166.5.129-47a6a2a8[0;37;40m", "[1;35;40mSIP/726002@acmetalk.net[0;37;40m") in new stack
 
[Ktelth9*CLI> 
We're at 217.14.132.185 port 18620
 
[Ktelth9*CLI> 
Answering/Requesting with root capability 8
 
[Ktelth9*CLI> 
Answering with preferred capability 0x4(ULAW)
 
[Ktelth9*CLI> 
Answering with preferred capability 0x400(ILBC)
 
[Ktelth9*CLI> 
Answering with preferred capability 0x2(GSM)
 
[Ktelth9*CLI> 
Answering with preferred capability 0x100(G729A)
 
[Ktelth9*CLI> 
Answering with non-codec capability 0x1(G723)
 
[Ktelth9*CLI> 
12 headers, 14 lines
 
[Ktelth9*CLI> 
Reliably Transmitting:
INVITE sip:726002@acmetalk.net SIP/2.0 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK15fba67f From: "07904306012" <sip:07904306012@217.14.132.162>;tag=as14f107a3 To: <sip:726002@acmetalk.net> Contact: <sip:07904306012@217.14.132.185> Call-ID: 1c64238d3d469d6e7a08cb3e6d5b29e2@217.14.132.162 CSeq: 102 INVITE User-Agent: acme VoIP Gateway Date: Fri, 18 Feb 2005 15:58:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 319  v=0 o=root 28176 28176 IN IP4 217.14.132.185 s=session c=IN IP4 217.14.132.185 t=0 0 m=audio 18620 RTP/AVP 8 0 97 3 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - -  (no NAT) to 217.14.132.178:5060
 
[Ktelth9*CLI> 
    -- Called 726002@acmetalk.net
 
[Ktelth9*CLI>  
Sip read: 
SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK15fba67f From: "07904306012" <sip:07904306012@217.14.132.162>;tag=as14f107a3 To: <sip:726002@acmetalk.net> Call-ID: 1c64238d3d469d6e7a08cb3e6d5b29e2@217.14.132.162 CSeq: 102 INVITE Server: Sip EXpress router (0.8.12-1rc6 (i386/linux)) Content-Length: 0 Warning: 392 217.14.132.178:5060 "Noisy feedback tells:  pid=17619 req_src_ip=217.14.132.185 req_src_port=5060 in_uri=sip:726002@acmetalk.net out_uri=sip:FIXytGX26r8RqmKcYcuK9f2cfg4S6TCuYcv-SKJCGQ.@217.14.132.172:5065;line=ecvb82yn via_cnt==1"  
 9 headers, 0 lines
 
[Ktelth9*CLI>  
Sip read: 
SIP/2.0 180 Ringing To: <sip:726002@acmetalk.net> From: "07904306012"<sip:07904306012@217.14.132.162>;tag=as14f107a3 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK15fba67f Call-ID: 1c64238d3d469d6e7a08cb3e6d5b29e2@217.14.132.162 CSeq: 102 INVITE Record-Route: <sip:726002@217.14.132.178;ftag=as14f107a3;lr=on> Contact: <sip:FQSiZLUmOm_BerJMsRtaJS1Mrt7COeB-sRsumHKSydGvWSG@217.14.132.172:5065;line=ecvb82yn> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0  
 11 headers, 0 lines
 
[Ktelth9*CLI> 
    -- SIP/acmetalk.net-7096 is ringing
 
[Ktelth9*CLI> 
Transmitting (NAT):
SIP/2.0 180 Ringing Via: SIP/2.0/UDP 213.166.5.129:5060;branch=z9hG4bK017f020b;received=213.166.5.129;rport=5060 From: "07904306012" <sip:07904306012@213.166.5.129>;tag=as2bce6bbd To: <sip:02070433288@incomingnum1.acmechat.org>;tag=as6ca4c6df Call-ID: 59d46e194f91c1c46a6857427b2d5c69@213.166.5.129 CSeq: 102 INVITE User-Agent: acme VoIP Gateway Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:02070433288@217.14.132.185> Content-Length: 0  
 to 213.166.5.129:5060
 
[Ktelth9*CLI>  
Sip read: 
SIP/2.0 180 Ringing To: <sip:726002@acmetalk.net> From: "07904306012"<sip:07904306012@217.14.132.162>;tag=as14f107a3 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK15fba67f Call-ID: 1c64238d3d469d6e7a08cb3e6d5b29e2@217.14.132.162 CSeq: 102 INVITE Record-Route: <sip:726002@217.14.132.178;ftag=as14f107a3;lr=on> Contact: <sip:FQSiZLUmOm_BerJMsRtaJS1Mrt7COeB-sRsumHKSydGvWSG@217.14.132.172:5065;line=ecvb82yn> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0  
 11 headers, 0 lines
 
[Ktelth9*CLI> 
    -- SIP/acmetalk.net-7096 is ringing
 
[Ktelth9*CLI>  
Sip read: 
SIP/2.0 180 Ringing To: <sip:726002@acmetalk.net> From: "07904306012"<sip:07904306012@217.14.132.162>;tag=as14f107a3 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK15fba67f Call-ID: 1c64238d3d469d6e7a08cb3e6d5b29e2@217.14.132.162 CSeq: 102 INVITE Record-Route: <sip:726002@217.14.132.178;ftag=as14f107a3;lr=on> Contact: <sip:FQSiZLUmOm_BerJMsRtaJS1Mrt7COeB-sRsumHKSydGvWSG@217.14.132.172:5065;line=ecvb82yn> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0  
 11 headers, 0 lines
 
[Ktelth9*CLI> 
    -- SIP/acmetalk.net-7096 is ringing
 
[Ktelth9*CLI>  
Sip read: 
SIP/2.0 200 Ok To: <sip:726002@acmetalk.net>;tag=434e0644 From: "07904306012"<sip:07904306012@217.14.132.162>;tag=as14f107a3 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK15fba67f Call-ID: 1c64238d3d469d6e7a08cb3e6d5b29e2@217.14.132.162 CSeq: 102 INVITE Record-Route: <sip:726002@217.14.132.178;ftag=as14f107a3;lr=on> Contact: <sip:FQSiZLUmOm_BerJMsRtaJS1Mrt7COeB-sRsumHKSydGvWSG@217.14.132.172:5065;line=ecvb82yn> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, MESSAGE, INFO Content-Type: application/sdp Supported: timer, replaces User-Agent: snom200-3.55 Allow-Events: talk, hold, refer Content-Length: 210  v=0 o=root 964039580 822375199 IN IP4 217.14.132.172 s=call c=IN IP4 217.14.132.172 t=0 0 m=audio 10062 RTP/AVP 8 101 a=fmtp:101 0-15 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv 
 14 headers, 10 lines
 Found RTP audio format 8
 Found RTP audio format 101
 Peer audio RTP is at port 217.14.132.172:10062
 Found description format pcma
 Found description format telephone-event
 Capabilities: us - 0x50e(GSM|ULAW|ALAW|G729A|ILBC), peer - audio=0x8(ALAW)/video=0x0(EMPTY), combined - 0x8(ALAW)
 Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723)
 list_route: hop: <sip:726002@217.14.132.178;ftag=as14f107a3;lr=on>
 list_route: hop: <sip:FQSiZLUmOm_BerJMsRtaJS1Mrt7COeB-sRsumHKSydGvWSG@217.14.132.172:5065;line=ecvb82yn>
 set_destination: Parsing <sip:726002@217.14.132.178;ftag=as14f107a3;lr=on> for address/port to send to
 set_destination: set destination to 217.14.132.178, port 5060
 Transmitting:
ACK sip:726002@acmetalk.net SIP/2.0 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK5bb85fb6 Route: <sip:FQSiZLUmOm_BerJMsRtaJS1Mrt7COeB-sRsumHKSydGvWSG@217.14.132.172:5065;line=ecvb82yn> From: "07904306012" <sip:07904306012@217.14.132.162>;tag=as14f107a3 To: <sip:726002@acmetalk.net>;tag=434e0644 Contact: <sip:07904306012@217.14.132.185> Call-ID: 1c64238d3d469d6e7a08cb3e6d5b29e2@217.14.132.162 CSeq: 102 ACK User-Agent: acme VoIP Gateway Content-Length: 0   (no NAT) to 217.14.132.178:5060
 
[Ktelth9*CLI> 
    -- SIP/acmetalk.net-7096 answered SIP/213.166.5.129-47a6a2a8
 
[Ktelth9*CLI> 
We're at 217.14.132.185 port 7062
 
[Ktelth9*CLI> 
Answering with preferred capability 0x8(ALAW)
 
[Ktelth9*CLI> 
Answering with preferred capability 0x4(ULAW)
 
[Ktelth9*CLI> 
Answering with preferred capability 0x400(ILBC)
 
[Ktelth9*CLI> 
Answering with preferred capability 0x2(GSM)
 
[Ktelth9*CLI> 
Answering with preferred capability 0x100(G729A)
 
[Ktelth9*CLI> 
Answering with non-codec capability 0x1(G723)
 
[Ktelth9*CLI> 
Reliably Transmitting (NAT):
SIP/2.0 200 OK Via: SIP/2.0/UDP 213.166.5.129:5060;branch=z9hG4bK017f020b;received=213.166.5.129;rport=5060 From: "07904306012" <sip:07904306012@213.166.5.129>;tag=as2bce6bbd To: <sip:02070433288@incomingnum1.acmechat.org>;tag=as6ca4c6df Call-ID: 59d46e194f91c1c46a6857427b2d5c69@213.166.5.129 CSeq: 102 INVITE User-Agent: acme VoIP Gateway Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:02070433288@217.14.132.185> Content-Type: application/sdp Content-Length: 318  v=0 o=root 28176 28176 IN IP4 217.14.132.185 s=session c=IN IP4 217.14.132.185 t=0 0 m=audio 7062 RTP/AVP 8 0 97 3 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - 
 to 213.166.5.129:5060
 
[Ktelth9*CLI> 
    -- Attempting native bridge of SIP/213.166.5.129-47a6a2a8 and SIP/acmetalk.net-7096
 
[Ktelth9*CLI> 
set_destination: Parsing <sip:726002@217.14.132.178;ftag=as14f107a3;lr=on> for address/port to send to
 
[Ktelth9*CLI> 
set_destination: set destination to 217.14.132.178, port 5060
 
[Ktelth9*CLI> 
We're at 217.14.132.185 port 18620
 
[Ktelth9*CLI> 
Answering/Requesting with root capability 8
 
[Ktelth9*CLI> 
Answering with preferred capability 0x4(ULAW)
 
[Ktelth9*CLI> 
Answering with preferred capability 0x2(GSM)
 
[Ktelth9*CLI> 
Answering with non-codec capability 0x1(G723)
 
[Ktelth9*CLI> 
12 headers, 12 lines
 
[Ktelth9*CLI> 
Reliably Transmitting:
INVITE sip:FQSiZLUmOm_BerJMsRtaJS1Mrt7COeB-sRsumHKSydGvWSG@217.14.132.172:5065 SIP/2.0 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK1a3628b4 Route: <sip:FQSiZLUmOm_BerJMsRtaJS1Mrt7COeB-sRsumHKSydGvWSG@217.14.132.172:5065;line=ecvb82yn> From: "07904306012" <sip:07904306012@217.14.132.162>;tag=as14f107a3 To: <sip:726002@acmetalk.net>;tag=434e0644 Contact: <sip:07904306012@217.14.132.185> Call-ID: 1c64238d3d469d6e7a08cb3e6d5b29e2@217.14.132.162 CSeq: 103 INVITE User-Agent: acme VoIP Gateway Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 265  v=0 o=root 28176 28177 IN IP4 213.166.5.129 s=session c=IN IP4 213.166.5.129 t=0 0 m=audio 21698 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - -  (no NAT) to 217.14.132.178:5060
 
[Ktelth9*CLI>  
Sip read: 
SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK1a3628b4 From: "07904306012" <sip:07904306012@217.14.132.162>;tag=as14f107a3 To: <sip:726002@acmetalk.net>;tag=434e0644 Call-ID: 1c64238d3d469d6e7a08cb3e6d5b29e2@217.14.132.162 CSeq: 103 INVITE Server: Sip EXpress router (0.8.12-1rc6 (i386/linux)) Content-Length: 0  
 8 headers, 0 lines
 
[Ktelth9*CLI>  
Sip read: 
ACK sip:02070433288@incomingnum1.acmechat.org SIP/2.0 Via: SIP/2.0/UDP 213.166.5.129:5060;branch=z9hG4bK45d5a321 From: "07904306012" <sip:07904306012@213.166.5.129>;tag=as2bce6bbd To: <sip:02070433288@incomingnum1.acmechat.org>;tag=as6ca4c6df Contact: <sip:07904306012@213.166.5.129> Call-ID: 59d46e194f91c1c46a6857427b2d5c69@213.166.5.129 CSeq: 102 ACK User-Agent: MSS VoIP Gateway Content-Length: 0  
 9 headers, 0 lines
 set_destination: Parsing <sip:07904306012@213.166.5.129> for address/port to send to
 set_destination: set destination to 213.166.5.129, port 5060
 We're at 217.14.132.185 port 7062
 Answering with preferred capability 0x8(ALAW)
 Answering with non-codec capability 0x1(G723)
 11 headers, 10 lines
 Reliably Transmitting:
INVITE sip:07904306012@213.166.5.129 SIP/2.0 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK0fbd95c6;rport From: <sip:02070433288@incomingnum1.acmechat.org>;tag=as6ca4c6df To: "07904306012" <sip:07904306012@213.166.5.129>;tag=as2bce6bbd Contact: <sip:02070433288@217.14.132.185> Call-ID: 59d46e194f91c1c46a6857427b2d5c69@213.166.5.129 CSeq: 102 INVITE User-Agent: acme VoIP Gateway Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 220  v=0 o=root 28176 28177 IN IP4 217.14.132.172 s=session c=IN IP4 217.14.132.172 t=0 0 m=audio 10062 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - -  (NAT) to 213.166.5.129:5060
 
[Ktelth9*CLI>  
Sip read: 
SIP/2.0 200 OK Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK0fbd95c6 From: <sip:02070433288@incomingnum1.acmechat.org>;tag=as6ca4c6df To: "07904306012" <sip:07904306012@213.166.5.129>;tag=as2bce6bbd Call-ID: 59d46e194f91c1c46a6857427b2d5c69@213.166.5.129 CSeq: 102 INVITE User-Agent: MSS VoIP Gateway Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:07904306012@213.166.5.129> Content-Type: application/sdp Content-Length: 265  v=0 o=root 30707 30708 IN IP4 213.166.5.129 s=session c=IN IP4 213.166.5.129 t=0 0 m=audio 21698 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - 
 11 headers, 12 lines
 Found RTP audio format 8
 Found RTP audio format 0
 Found RTP audio format 3
 Found RTP audio format 101
 Peer audio RTP is at port 213.166.5.129:21698
 Found description format PCMA
 Found description format PCMU
 Found description format GSM
 Found description format telephone-event
 Capabilities: us - 0x50e(GSM|ULAW|ALAW|G729A|ILBC), peer - audio=0xe(GSM|ULAW|ALAW)/video=0x0(EMPTY), combined - 0xe(GSM|ULAW|ALAW)
 Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723)
 list_route: hop: <sip:07904306012@213.166.5.129>
 set_destination: Parsing <sip:07904306012@213.166.5.129> for address/port to send to
 set_destination: set destination to 213.166.5.129, port 5060
 Transmitting:
ACK sip:07904306012@213.166.5.129 SIP/2.0 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK02cde259;rport From: <sip:02070433288@incomingnum1.acmechat.org>;tag=as6ca4c6df To: "07904306012" <sip:07904306012@213.166.5.129>;tag=as2bce6bbd Contact: <sip:02070433288@217.14.132.185> Call-ID: 59d46e194f91c1c46a6857427b2d5c69@213.166.5.129 CSeq: 102 ACK User-Agent: acme VoIP Gateway Content-Length: 0   (NAT) to 213.166.5.129:5060
 
[Ktelth9*CLI>  
Sip read: 
SIP/2.0 200 Ok To: <sip:726002@acmetalk.net>;tag=434e0644 From: "07904306012"<sip:07904306012@217.14.132.162>;tag=as14f107a3 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK1a3628b4 Call-ID: 1c64238d3d469d6e7a08cb3e6d5b29e2@217.14.132.162 CSeq: 103 INVITE Record-Route: <sip:FQSiZLUmOm_BerJMsRtaJS1Mrt7COeB-sRsumHKSydGvWSG@217.14.132.178;ftag=as14f107a3;lr=on> Contact: <sip:FQSiZLUmOm_BerJMsRtaJS1Mrt7COeB-sRsumHKSydGvWSG@217.14.132.172:5065;line=ecvb82yn> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, MESSAGE, INFO Content-Type: application/sdp Supported: timer, replaces User-Agent: snom200-3.55 Allow-Events: talk, hold, refer Content-Length: 210  v=0 o=root 964039580 822375200 IN IP4 217.14.132.172 s=call c=IN IP4 217.14.132.172 t=0 0 m=audio 10062 RTP/AVP 8 101 a=fmtp:101 0-15 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv 
 14 headers, 10 lines
 Found RTP audio format 8
 Found RTP audio format 101
 Peer audio RTP is at port 217.14.132.172:10062
 Found description format pcma
 Found description format telephone-event
 Capabilities: us - 0x50e(GSM|ULAW|ALAW|G729A|ILBC), peer - audio=0x8(ALAW)/video=0x0(EMPTY), combined - 0x8(ALAW)
 Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723)
 set_destination: Parsing <sip:726002@217.14.132.178;ftag=as14f107a3;lr=on> for address/port to send to
 set_destination: set destination to 217.14.132.178, port 5060
 Transmitting:
ACK sip:FQSiZLUmOm_BerJMsRtaJS1Mrt7COeB-sRsumHKSydGvWSG@217.14.132.172:5065 SIP/2.0 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK4908a0d1 Route: <sip:FQSiZLUmOm_BerJMsRtaJS1Mrt7COeB-sRsumHKSydGvWSG@217.14.132.172:5065;line=ecvb82yn> From: "07904306012" <sip:07904306012@217.14.132.162>;tag=as14f107a3 To: <sip:726002@acmetalk.net>;tag=434e0644 Contact: <sip:07904306012@217.14.132.185> Call-ID: 1c64238d3d469d6e7a08cb3e6d5b29e2@217.14.132.162 CSeq: 103 ACK User-Agent: acme VoIP Gateway Content-Length: 0   (no NAT) to 217.14.132.178:5060
 
[Ktelth9*CLI>  
Sip read: 
BYE sip:02070438202@217.14.132.185 SIP/2.0 Via: SIP/2.0/UDP 213.166.5.129:5060;branch=z9hG4bK3d44ddd3 From: "02076190123" <sip:02076190123@213.166.5.129>;tag=as08720ecd To: <sip:02070438202@incomingnum1.acmechat.org>;tag=as4ad1b15f Contact: <sip:02076190123@213.166.5.129> Call-ID: 02e971a947eae9bd2695aa230f351060@213.166.5.129 CSeq: 103 BYE User-Agent: MSS VoIP Gateway Content-Length: 0  
 9 headers, 0 lines
 Sending to 213.166.5.129 : 5060 (NAT)
 Transmitting (NAT):
SIP/2.0 200 OK 
[Ktelth9*CLI> 
Via: SIP/2.0/UDP 213.166.5.129:5060;branch=z9hG4bK3d44ddd3;received=213.166.5.129;rport=5060 From: "02076190123" <sip:02076190123@213.166.5.129>;tag=as08720ecd To: <sip:02070438202@incomingnum1.acmechat.org>;tag=as4ad1b15f Call-ID: 02e971a947eae9bd2695aa230f351060@213.166.5.129 CSeq: 103 BYE User-Agent: acme VoIP Gateway Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:02070438202@217.14.132.185> Content-Length: 0  
 to 213.166.5.129:5060
 
[Ktelth9*CLI> 
    -- Hungup 'IAX2/83.216.129.33:4569/9'
   == Spawn extension (prepaid-dial, s, 1) exited non-zero on 'SIP/213.166.5.129-47a11328'
 
[Ktelth9*CLI> 
Feb 18 15:58:52 [31;40mERROR[0;37;40m[28110]: [1;37;40mcdr_addon_mysql.c[0;37;40m:[1;37;40m122[0;37;40m [1;37;40mmysql_log[0;37;40m:  cdr_mysql: cannot connect to database server localhost.  Call will not be logged
 
[Ktelth9*CLI> 
Destroying call '02e971a947eae9bd2695aa230f351060@213.166.5.129'
 
[Ktelth9*CLI> 
11 headers, 0 lines
 Reliably Transmitting:
REGISTER sip:iax3.hitchhiker.net SIP/2.0 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK2299a7f1 From: <sip:acme@iax3.hitchhiker.net>;tag=as471b57ff To: <sip:acme@iax3.hitchhiker.net> Call-ID: 44a08a3f44d83edf7ebecfb2138abe69@217.14.132.162 CSeq: 112 REGISTER User-Agent: acme VoIP Gateway Expires: 120 Contact: <sip:s@217.14.132.185> Event: registration Content-Length: 0   (no NAT) to 213.166.5.130:5060
 
[Ktelth9*CLI>  
Sip read: 
SIP/2.0 100 Trying Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK2299a7f1 From: <sip:acme@iax3.hitchhiker.net>;tag=as471b57ff To: <sip:acme@iax3.hitchhiker.net>;tag=as0f6a0c4c Call-ID: 44a08a3f44d83edf7ebecfb2138abe69@217.14.132.162 CSeq: 112 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:acme@213.166.5.130> Content-Length: 0  
 10 headers, 0 lines
 
[Ktelth9*CLI>  
Sip read: 
SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK2299a7f1 From: <sip:acme@iax3.hitchhiker.net>;tag=as471b57ff To: <sip:acme@iax3.hitchhiker.net>;tag=as0f6a0c4c Call-ID: 44a08a3f44d83edf7ebecfb2138abe69@217.14.132.162 CSeq: 112 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:acme@213.166.5.130> WWW-Authenticate: Digest realm="asterisk", nonce="0097ba25" Content-Length: 0  
 11 headers, 0 lines
 12 headers, 0 lines
 Reliably Transmitting:
REGISTER sip:iax3.hitchhiker.net SIP/2.0 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK29edaf76 From: <sip:acme@iax3.hitchhiker.net>;tag=as05c4af94 To: <sip:acme@iax3.hitchhiker.net> Call-ID: 44a08a3f44d83edf7ebecfb2138abe69@217.14.132.162 CSeq: 113 REGISTER User-Agent: acme VoIP Gateway Authorization: Digest username="acme", realm="asterisk", algorithm=MD5, uri="sip:acme@213.166.5.130", nonce="0097ba25", response="91b1c62570c65fdaa68ae309e6870447", opaque="" Expires: 120 Contact: <sip:s@217.14.132.185> Event: registration Content-Length: 0   (no NAT) to 213.166.5.130:5060
 
[Ktelth9*CLI>  
Sip read: 
SIP/2.0 100 Trying Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK29edaf76 From: <sip:acme@iax3.hitchhiker.net>;tag=as05c4af94 To: <sip:acme@iax3.hitchhiker.net>;tag=as0f6a0c4c Call-ID: 44a08a3f44d83edf7ebecfb2138abe69@217.14.132.162 CSeq: 113 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:acme@213.166.5.130> Content-Length: 0  
 10 headers, 0 lines
 
[Ktelth9*CLI>  
Sip read: 
BYE sip:02070433288@217.14.132.185 SIP/2.0 Via: SIP/2.0/UDP 213.166.5.129:5060;branch=z9hG4bK0c96a8a6 From: "07904306012" <sip:07904306012@213.166.5.129>;tag=as2bce6bbd To: <sip:02070433288@incomingnum1.acmechat.org>;tag=as6ca4c6df Contact: <sip:07904306012@213.166.5.129> Call-ID: 59d46e194f91c1c46a6857427b2d5c69@213.166.5.129 CSeq: 103 BYE User-Agent: MSS VoIP Gateway Content-Length: 0  
 9 headers, 0 lines
 Sending to 213.166.5.129 : 5060 (NAT)
 Transmitting (NAT):
SIP/2.0 200 OK 
[Ktelth9*CLI> 
Via: SIP/2.0/UDP 213.166.5.129:5060;branch=z9hG4bK0c96a8a6;received=213.166.5.129;rport=5060 From: "07904306012" <sip:07904306012@213.166.5.129>;tag=as2bce6bbd To: <sip:02070433288@incomingnum1.acmechat.org>;tag=as6ca4c6df Call-ID: 59d46e194f91c1c46a6857427b2d5c69@213.166.5.129 CSeq: 103 BYE User-Agent: acme VoIP Gateway Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:02070433288@217.14.132.185> Content-Length: 0  
 to 213.166.5.129:5060
 
[Ktelth9*CLI> 
set_destination: Parsing <sip:726002@217.14.132.178;ftag=as14f107a3;lr=on> for address/port to send to
 set_destination: set destination to 217.14.132.178, port 5060
 We're at 217.14.132.185 port 18620
 Answering/Requesting with root capability 8
 Answering with preferred capability 0x4(ULAW)
 Answering with preferred capability 0x400(ILBC)
 Answering with preferred capability 0x2(GSM)
 Answering with preferred capability 0x100(G729A)
 Answering with non-codec capability 0x1(G723)
 12 headers, 14 lines
 Reliably Transmitting:
INVITE sip:FQSiZLUmOm_BerJMsRtaJS1Mrt7COeB-sRsumHKSydGvWSG@217.14.132.172:5065 SIP/2.0 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK2cd029a7 Route: <sip:FQSiZLUmOm_BerJMsRtaJS1Mrt7COeB-sRsumHKSydGvWSG@217.14.132.172:5065;line=ecvb82yn> From: "07904306012" <sip:07904306012@217.14.132.162>;tag=as14f107a3 To: <sip:726002@acmetalk.net>;tag=434e0644 Contact: <sip:07904306012@217.14.132.185> Call-ID: 1c64238d3d469d6e7a08cb3e6d5b29e2@217.14.132.162 CSeq: 104 INVITE User-Agent: acme VoIP Gateway Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 319  v=0 o=root 28176 28178 IN IP4 217.14.132.185 s=session c=IN IP4 217.14.132.185 t=0 0 m=audio 18620 RTP/AVP 8 0 97 3 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - -  (no NAT) to 217.14.132.178:5060
 
[Ktelth9*CLI>  
Sip read: 
SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK2cd029a7 From: "07904306012" <sip:07904306012@217.14.132.162>;tag=as14f107a3 To: <sip:726002@acmetalk.net>;tag=434e0644 Call-ID: 1c64238d3d469d6e7a08cb3e6d5b29e2@217.14.132.162 CSeq: 104 INVITE Server: Sip EXpress router (0.8.12-1rc6 (i386/linux)) Content-Length: 0  
 8 headers, 0 lines
   == Spawn extension (prepaid-dial, s, 1) exited non-zero on 'SIP/213.166.5.129-47a6a2a8'
 
[Ktelth9*CLI> 
Feb 18 15:58:53 [31;40mERROR[0;37;40m[28176]: [1;37;40mcdr_addon_mysql.c[0;37;40m:[1;37;40m122[0;37;40m [1;37;40mmysql_log[0;37;40m:  cdr_mysql: cannot connect to database server localhost.  Call will not be logged
 
[Ktelth9*CLI>  
Sip read: 
SIP/2.0 200 Ok To: <sip:726002@acmetalk.net>;tag=434e0644 From: "07904306012"<sip:07904306012@217.14.132.162>;tag=as14f107a3 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK2cd029a7 Call-ID: 1c64238d3d469d6e7a08cb3e6d5b29e2@217.14.132.162 CSeq: 104 INVITE Record-Route: <sip:FQSiZLUmOm_BerJMsRtaJS1Mrt7COeB-sRsumHKSydGvWSG@217.14.132.178;ftag=as14f107a3;lr=on> Contact: <sip:FQSiZLUmOm_BerJMsRtaJS1Mrt7COeB-sRsumHKSydGvWSG@217.14.132.172:5065;line=ecvb82yn> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, MESSAGE, INFO Content-Type: application/sdp Supported: timer, replaces User-Agent: snom200-3.55 Allow-Events: talk, hold, refer Content-Length: 210  v=0 o=root 964039580 822375201 IN IP4 217.14.132.172 s=call c=IN IP4 217.14.132.172 t=0 0 m=audio 10062 RTP/AVP 8 101 a=fmtp:101 0-15 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv 
 14 headers, 10 lines
 Found RTP audio format 8
 Found RTP audio format 101
 Peer audio RTP is at port 217.14.132.172:10062
 Found description format pcma
 Found description format telephone-event
 Capabilities: us - 0x50e(GSM|ULAW|ALAW|G729A|ILBC), peer - audio=0x8(ALAW)/video=0x0(EMPTY), combined - 0x8(ALAW)
 Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723)
 set_destination: Parsing <sip:726002@217.14.132.178;ftag=as14f107a3;lr=on> for address/port to send to
 set_destination: set destination to 217.14.132.178, port 5060
 Transmitting:
ACK sip:FQSiZLUmOm_BerJMsRtaJS1Mrt7COeB-sRsumHKSydGvWSG@217.14.132.172:5065 SIP/2.0 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK6bd02b5c Route: <sip:FQSiZLUmOm_BerJMsRtaJS1Mrt7COeB-sRsumHKSydGvWSG@217.14.132.172:5065;line=ecvb82yn> From: "07904306012" <sip:07904306012@217.14.132.162>;tag=as14f107a3 To: <sip:726002@acmetalk.net>;tag=434e0644 Contact: <sip:07904306012@217.14.132.185> Call-ID: 1c64238d3d469d6e7a08cb3e6d5b29e2@217.14.132.162 CSeq: 104 ACK User-Agent: acme VoIP Gateway Content-Length: 0   (no NAT) to 217.14.132.178:5060
 set_destination: Parsing <sip:726002@217.14.132.178;ftag=as14f107a3;lr=on> for address/port to send to
 set_destination: set destination to 217.14.132.178, port 5060
 Reliably Transmitting:
BYE sip:FQSiZLUmOm_BerJMsRtaJS1Mrt7COeB-sRsumHKSydGvWSG@217.14.132.172:5065 SIP/2.0 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK4bb53b76 Route: <sip:FQSiZLUmOm_BerJMsRtaJS1Mrt7COeB-sRsumHKSydGvWSG@217.14.132.172:5065;line=ecvb82yn> From: "07904306012" <sip:07904306012@217.14.132.162>;tag=as14f107a3 To: <sip:726002@acmetalk.net>;tag=434e0644 Contact: <sip:07904306012@217.14.132.185> Call-ID: 1c64238d3d469d6e7a08cb3e6d5b29e2@217.14.132.162 CSeq: 105 BYE User-Agent: acme VoIP Gateway Content-Length: 0   (no NAT) to 217.14.132.178:5060
 Destroying call '59d46e194f91c1c46a6857427b2d5c69@213.166.5.129'
 
[Ktelth9*CLI>  
Sip read: 
SIP/2.0 200 OK Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK29edaf76 From: <sip:acme@iax3.hitchhiker.net>;tag=as05c4af94 To: <sip:acme@iax3.hitchhiker.net>;tag=as0f6a0c4c Call-ID: 44a08a3f44d83edf7ebecfb2138abe69@217.14.132.162 CSeq: 113 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 120 Contact: <sip:s@217.14.132.185>;expires=120 Date: Fri, 18 Feb 2005 15:52:01 GMT Content-Length: 0  
 12 headers, 0 lines
 Destroying call '44a08a3f44d83edf7ebecfb2138abe69@217.14.132.162'
 
[Ktelth9*CLI>  
Sip read: 
SIP/2.0 200 OK To: <sip:726002@acmetalk.net>;tag=434e0644 From: "07904306012"<sip:07904306012@217.14.132.162>;tag=as14f107a3 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK4bb53b76 Call-ID: 1c64238d3d469d6e7a08cb3e6d5b29e2@217.14.132.162 CSeq: 105 BYE Record-Route: <sip:FQSiZLUmOm_BerJMsRtaJS1Mrt7COeB-sRsumHKSydGvWSG@217.14.132.178;ftag=as14f107a3;lr=on> Contact: <sip:FQSiZLUmOm_BerJMsRtaJS1Mrt7COeB-sRsumHKSydGvWSG@217.14.132.172:5065;line=ecvb82yn> User-Agent: snom200-3.55 Content-Length: 0  
 10 headers, 0 lines
 Destroying call '1c64238d3d469d6e7a08cb3e6d5b29e2@217.14.132.162'
 
[Ktelth9*CLI>  
Sip read: 
INVITE sip:02070433288@incomingnum1.acmechat.org SIP/2.0 Via: SIP/2.0/UDP 213.166.5.129:5060;branch=z9hG4bK253776af From: "07904306012" <sip:07904306012@213.166.5.129>;tag=as3d63a8f3 To: <sip:02070433288@incomingnum1.acmechat.org> Contact: <sip:07904306012@213.166.5.129> Call-ID: 72bd80002f592b26510108e275c6d18c@213.166.5.129 CSeq: 102 INVITE User-Agent: MSS VoIP Gateway Date: Fri, 18 Feb 2005 15:52:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 265  v=0 o=root 30728 30728 IN IP4 213.166.5.129 s=session c=IN IP4 213.166.5.129 t=0 0 m=audio 10790 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - 
 12 headers, 12 lines
 Using latest request as basis request
 Sending to 213.166.5.129 : 5060 (NAT)
 Found RTP audio format 8
 Found RTP audio format 0
 Found RTP audio format 3
 Found RTP audio format 101
 Peer audio RTP is at port 213.166.5.129:10790
 Found description format PCMA
 Found description format PCMU
 Found description format GSM
 Found description format telephone-event
 Capabilities: us - 0x50e(GSM|ULAW|ALAW|G729A|ILBC), peer - audio=0xe(GSM|ULAW|ALAW)/video=0x0(EMPTY), combined - 0xe(GSM|ULAW|ALAW)
 Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723)
 Found no matching peer or user for '213.166.5.129:5060'
 Looking for 02070433288 in prepaid
 list_route: hop: <sip:07904306012@213.166.5.129>
 Transmitting (NAT):
SIP/2.0 100 Trying Via: SIP/2.0/UDP 213.166.5.129:5060;branch=z9hG4bK253776af;received=213.166.5.129;rport=5060 From: "07904306012" <sip:07904306012@213.166.5.129>;tag=as3d63a8f3 To: <sip:02070433288@incomingnum1.acmechat.org>;tag=as0bdcc570 Call-ID: 72bd80002f592b26510108e275c6d18c@213.166.5.129 CSeq: 102 INVITE User-Agent: acme VoIP Gateway Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:02070433288@217.14.132.185> Content-Length: 0  
 to 213.166.5.129:5060
 
[Ktelth9*CLI> 
    -- Executing [1;36;40mAGI[0;37;40m("[1;35;40mSIP/213.166.5.129-47a6a2a8[0;37;40m", "[1;35;40mlookup_dialplan.pl|02070433288[0;37;40m") in new stack
 
[Ktelth9*CLI> 
    -- Launched AGI Script /var/lib/asterisk/agi-bin/lookup_dialplan.pl
 
[Ktelth9*CLI> 
    -- AGI Script lookup_dialplan.pl completed, returning 0
 
[Ktelth9*CLI> 
    -- Executing [1;36;40mDial[0;37;40m("[1;35;40mSIP/213.166.5.129-47a6a2a8[0;37;40m", "[1;35;40mSIP/726002@acmetalk.net[0;37;40m") in new stack
 
[Ktelth9*CLI> 
We're at 217.14.132.185 port 9640
 
[Ktelth9*CLI> 
Answering/Requesting with root capability 8
 
[Ktelth9*CLI> 
Answering with preferred capability 0x4(ULAW)
 
[Ktelth9*CLI> 
Answering with preferred capability 0x400(ILBC)
 
[Ktelth9*CLI> 
Answering with preferred capability 0x2(GSM)
 
[Ktelth9*CLI> 
Answering with preferred capability 0x100(G729A)
 
[Ktelth9*CLI> 
Answering with non-codec capability 0x1(G723)
 
[Ktelth9*CLI> 
12 headers, 14 lines
 
[Ktelth9*CLI> 
Reliably Transmitting:
INVITE sip:726002@acmetalk.net SIP/2.0 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK386fc4ec From: "07904306012" <sip:07904306012@217.14.132.162>;tag=as3dbf7d16 To: <sip:726002@acmetalk.net> Contact: <sip:07904306012@217.14.132.185> Call-ID: 6182fd661973ee7e02da14820fcfef51@217.14.132.162 CSeq: 102 INVITE User-Agent: acme VoIP Gateway Date: Fri, 18 Feb 2005 15:59:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 318  v=0 o=root 28181 28181 IN IP4 217.14.132.185 s=session c=IN IP4 217.14.132.185 t=0 0 m=audio 9640 RTP/AVP 8 0 97 3 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - -  (no NAT) to 217.14.132.178:5060
 
[Ktelth9*CLI> 
    -- Called 726002@acmetalk.net
 
[Ktelth9*CLI>  
Sip read: 
SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK386fc4ec From: "07904306012" <sip:07904306012@217.14.132.162>;tag=as3dbf7d16 To: <sip:726002@acmetalk.net> Call-ID: 6182fd661973ee7e02da14820fcfef51@217.14.132.162 CSeq: 102 INVITE Server: Sip EXpress router (0.8.12-1rc6 (i386/linux)) Content-Length: 0 Warning: 392 217.14.132.178:5060 "Noisy feedback tells:  pid=17628 req_src_ip=217.14.132.185 req_src_port=5060 in_uri=sip:726002@acmetalk.net out_uri=sip:FIXytGX26r8RqmKcYcuK9f2cfg4S6TCuYcv-SKJCGQ.@217.14.132.172:5065;line=ecvb82yn via_cnt==1"  
 9 headers, 0 lines
 
[Ktelth9*CLI>  
Sip read: 
SIP/2.0 180 Ringing To: <sip:726002@acmetalk.net> From: "07904306012"<sip:07904306012@217.14.132.162>;tag=as3dbf7d16 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK386fc4ec Call-ID: 6182fd661973ee7e02da14820fcfef51@217.14.132.162 CSeq: 102 INVITE Record-Route: <sip:726002@217.14.132.178;ftag=as3dbf7d16;lr=on> Contact: <sip:FQSiZLUmOm_BerJMsRtaJS1Mrt7COeB-sRsumHKSydGvWSG@217.14.132.172:5065;line=ecvb82yn> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0  
 11 headers, 0 lines
 
[Ktelth9*CLI> 
    -- SIP/acmetalk.net-be80 is ringing
 
[Ktelth9*CLI> 
Transmitting (NAT):
SIP/2.0 180 Ringing Via: SIP/2.0/UDP 213.166.5.129:5060;branch=z9hG4bK253776af;received=213.166.5.129;rport=5060 From: "07904306012" <sip:07904306012@213.166.5.129>;tag=as3d63a8f3 To: <sip:02070433288@incomingnum1.acmechat.org>;tag=as0bdcc570 Call-ID: 72bd80002f592b26510108e275c6d18c@213.166.5.129 CSeq: 102 INVITE User-Agent: acme VoIP Gateway Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:02070433288@217.14.132.185> Content-Length: 0  
 to 213.166.5.129:5060
 
[Ktelth9*CLI>  
Sip read: 
SIP/2.0 180 Ringing To: <sip:726002@acmetalk.net> From: "07904306012"<sip:07904306012@217.14.132.162>;tag=as3dbf7d16 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK386fc4ec Call-ID: 6182fd661973ee7e02da14820fcfef51@217.14.132.162 CSeq: 102 INVITE Record-Route: <sip:726002@217.14.132.178;ftag=as3dbf7d16;lr=on> Contact: <sip:FQSiZLUmOm_BerJMsRtaJS1Mrt7COeB-sRsumHKSydGvWSG@217.14.132.172:5065;line=ecvb82yn> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0  
 11 headers, 0 lines
 
[Ktelth9*CLI> 
    -- SIP/acmetalk.net-be80 is ringing
 
[Ktelth9*CLI>  
Sip read: 
SIP/2.0 180 Ringing To: <sip:726002@acmetalk.net> From: "07904306012"<sip:07904306012@217.14.132.162>;tag=as3dbf7d16 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK386fc4ec Call-ID: 6182fd661973ee7e02da14820fcfef51@217.14.132.162 CSeq: 102 INVITE Record-Route: <sip:726002@217.14.132.178;ftag=as3dbf7d16;lr=on> Contact: <sip:FQSiZLUmOm_BerJMsRtaJS1Mrt7COeB-sRsumHKSydGvWSG@217.14.132.172:5065;line=ecvb82yn> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0  
 11 headers, 0 lines
 
[Ktelth9*CLI> 
    -- SIP/acmetalk.net-be80 is ringing
 
[Ktelth9*CLI>  
Sip read: 
SIP/2.0 180 Ringing To: <sip:726002@acmetalk.net> From: "07904306012"<sip:07904306012@217.14.132.162>;tag=as3dbf7d16 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK386fc4ec Call-ID: 6182fd661973ee7e02da14820fcfef51@217.14.132.162 CSeq: 102 INVITE Record-Route: <sip:726002@217.14.132.178;ftag=as3dbf7d16;lr=on> Contact: <sip:FQSiZLUmOm_BerJMsRtaJS1Mrt7COeB-sRsumHKSydGvWSG@217.14.132.172:5065;line=ecvb82yn> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0  
 11 headers, 0 lines
 
[Ktelth9*CLI> 
    -- SIP/acmetalk.net-be80 is ringing
 
[Ktelth9*CLI>  
Sip read: 
SIP/2.0 200 Ok To: <sip:726002@acmetalk.net>;tag=7ae6ab74 From: "07904306012"<sip:07904306012@217.14.132.162>;tag=as3dbf7d16 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK386fc4ec Call-ID: 6182fd661973ee7e02da14820fcfef51@217.14.132.162 CSeq: 102 INVITE Record-Route: <sip:726002@217.14.132.178;ftag=as3dbf7d16;lr=on> Contact: <sip:FQSiZLUmOm_BerJMsRtaJS1Mrt7COeB-sRsumHKSydGvWSG@217.14.132.172:5065;line=ecvb82yn> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, MESSAGE, INFO Content-Type: application/sdp Supported: timer, replaces User-Agent: snom200-3.55 Allow-Events: talk, hold, refer Content-Length: 210  v=0 o=root 138988592 195500906 IN IP4 217.14.132.172 s=call c=IN IP4 217.14.132.172 t=0 0 m=audio 10066 RTP/AVP 8 101 a=fmtp:101 0-15 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv 
 14 headers, 10 lines
 Found RTP audio format 8
 Found RTP audio format 101
 Peer audio RTP is at port 217.14.132.172:10066
 Found description format pcma
 Found description format telephone-event
 Capabilities: us - 0x50e(GSM|ULAW|ALAW|G729A|ILBC), peer - audio=0x8(ALAW)/video=0x0(EMPTY), combined - 0x8(ALAW)
 Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723)
 list_route: hop: <sip:726002@217.14.132.178;ftag=as3dbf7d16;lr=on>
 list_route: hop: <sip:FQSiZLUmOm_BerJMsRtaJS1Mrt7COeB-sRsumHKSydGvWSG@217.14.132.172:5065;line=ecvb82yn>
 set_destination: Parsing <sip:726002@217.14.132.178;ftag=as3dbf7d16;lr=on> for address/port to send to
 set_destination: set destination to 217.14.132.178, port 5060
 Transmitting:
ACK sip:726002@acmetalk.net SIP/2.0 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK78c85bb8 Route: <sip:FQSiZLUmOm_BerJMsRtaJS1Mrt7COeB-sRsumHKSydGvWSG@217.14.132.172:5065;line=ecvb82yn> From: "07904306012" <sip:07904306012@217.14.132.162>;tag=as3dbf7d16 To: <sip:726002@acmetalk.net>;tag=7ae6ab74 Contact: <sip:07904306012@217.14.132.185> Call-ID: 6182fd661973ee7e02da14820fcfef51@217.14.132.162 CSeq: 102 ACK User-Agent: acme VoIP Gateway Content-Length: 0   (no NAT) to 217.14.132.178:5060
 
[Ktelth9*CLI> 
    -- SIP/acmetalk.net-be80 answered SIP/213.166.5.129-47a6a2a8
 
[Ktelth9*CLI> 
We're at 217.14.132.185 port 9816
 
[Ktelth9*CLI> 
Answering with preferred capability 0x8(ALAW)
 
[Ktelth9*CLI> 
Answering with preferred capability 0x4(ULAW)
 
[Ktelth9*CLI> 
Answering with preferred capability 0x400(ILBC)
 
[Ktelth9*CLI> 
Answering with preferred capability 0x2(GSM)
 
[Ktelth9*CLI> 
Answering with preferred capability 0x100(G729A)
 
[Ktelth9*CLI> 
Answering with non-codec capability 0x1(G723)
 
[Ktelth9*CLI> 
Reliably Transmitting (NAT):
SIP/2.0 200 OK Via: SIP/2.0/UDP 213.166.5.129:5060;branch=z9hG4bK253776af;received=213.166.5.129;rport=5060 From: "07904306012" <sip:07904306012@213.166.5.129>;tag=as3d63a8f3 To: <sip:02070433288@incomingnum1.acmechat.org>;tag=as0bdcc570 Call-ID: 72bd80002f592b26510108e275c6d18c@213.166.5.129 CSeq: 102 INVITE User-Agent: acme VoIP Gateway Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:02070433288@217.14.132.185> Content-Type: application/sdp Content-Length: 318  v=0 o=root 28181 28181 IN IP4 217.14.132.185 s=session c=IN IP4 217.14.132.185 t=0 0 m=audio 9816 RTP/AVP 8 0 97 3 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - 
 to 213.166.5.129:5060
 
[Ktelth9*CLI> 
    -- Attempting native bridge of SIP/213.166.5.129-47a6a2a8 and SIP/acmetalk.net-be80
 
[Ktelth9*CLI> 
set_destination: Parsing <sip:726002@217.14.132.178;ftag=as3dbf7d16;lr=on> for address/port to send to
 
[Ktelth9*CLI> 
set_destination: set destination to 217.14.132.178, port 5060
 
[Ktelth9*CLI> 
We're at 217.14.132.185 port 9640
 
[Ktelth9*CLI> 
Answering/Requesting with root capability 8
 
[Ktelth9*CLI> 
Answering with preferred capability 0x4(ULAW)
 
[Ktelth9*CLI> 
Answering with preferred capability 0x2(GSM)
 
[Ktelth9*CLI> 
Answering with non-codec capability 0x1(G723)
 
[Ktelth9*CLI> 
12 headers, 12 lines
 
[Ktelth9*CLI> 
Reliably Transmitting:
INVITE sip:FQSiZLUmOm_BerJMsRtaJS1Mrt7COeB-sRsumHKSydGvWSG@217.14.132.172:5065 SIP/2.0 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK5adc4dc7 Route: <sip:FQSiZLUmOm_BerJMsRtaJS1Mrt7COeB-sRsumHKSydGvWSG@217.14.132.172:5065;line=ecvb82yn> From: "07904306012" <sip:07904306012@217.14.132.162>;tag=as3dbf7d16 To: <sip:726002@acmetalk.net>;tag=7ae6ab74 Contact: <sip:07904306012@217.14.132.185> Call-ID: 6182fd661973ee7e02da14820fcfef51@217.14.132.162 CSeq: 103 INVITE User-Agent: acme VoIP Gateway Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 265  v=0 o=root 28181 28182 IN IP4 213.166.5.129 s=session c=IN IP4 213.166.5.129 t=0 0 m=audio 10790 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - -  (no NAT) to 217.14.132.178:5060
 
[Ktelth9*CLI>  
Sip read: 
SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK5adc4dc7 From: "07904306012" <sip:07904306012@217.14.132.162>;tag=as3dbf7d16 To: <sip:726002@acmetalk.net>;tag=7ae6ab74 Call-ID: 6182fd661973ee7e02da14820fcfef51@217.14.132.162 CSeq: 103 INVITE Server: Sip EXpress router (0.8.12-1rc6 (i386/linux)) Content-Length: 0  
 8 headers, 0 lines
 
[Ktelth9*CLI>  
Sip read: 
ACK sip:02070433288@incomingnum1.acmechat.org SIP/2.0 Via: SIP/2.0/UDP 213.166.5.129:5060;branch=z9hG4bK63133bfa From: "07904306012" <sip:07904306012@213.166.5.129>;tag=as3d63a8f3 To: <sip:02070433288@incomingnum1.acmechat.org>;tag=as0bdcc570 Contact: <sip:07904306012@213.166.5.129> Call-ID: 72bd80002f592b26510108e275c6d18c@213.166.5.129 CSeq: 102 ACK User-Agent: MSS VoIP Gateway Content-Length: 0  
 9 headers, 0 lines
 set_destination: Parsing <sip:07904306012@213.166.5.129> for address/port to send to
 set_destination: set destination to 213.166.5.129, port 5060
 We're at 217.14.132.185 port 9816
 Answering with preferred capability 0x8(ALAW)
 Answering with non-codec capability 0x1(G723)
 11 headers, 10 lines
 Reliably Transmitting:
INVITE sip:07904306012@213.166.5.129 SIP/2.0 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK71fcf293;rport From: <sip:02070433288@incomingnum1.acmechat.org>;tag=as0bdcc570 To: "07904306012" <sip:07904306012@213.166.5.129>;tag=as3d63a8f3 Contact: <sip:02070433288@217.14.132.185> Call-ID: 72bd80002f592b26510108e275c6d18c@213.166.5.129 CSeq: 102 INVITE User-Agent: acme VoIP Gateway Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 220  v=0 o=root 28181 28182 IN IP4 217.14.132.172 s=session c=IN IP4 217.14.132.172 t=0 0 m=audio 10066 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - -  (NAT) to 213.166.5.129:5060
 
[Ktelth9*CLI>  
Sip read: 
SIP/2.0 200 OK Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK71fcf293 From: <sip:02070433288@incomingnum1.acmechat.org>;tag=as0bdcc570 To: "07904306012" <sip:07904306012@213.166.5.129>;tag=as3d63a8f3 Call-ID: 72bd80002f592b26510108e275c6d18c@213.166.5.129 CSeq: 102 INVITE User-Agent: MSS VoIP Gateway Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:07904306012@213.166.5.129> Content-Type: application/sdp Content-Length: 265  v=0 o=root 30728 30729 IN IP4 213.166.5.129 s=session c=IN IP4 213.166.5.129 t=0 0 m=audio 10790 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - 
 11 headers, 12 lines
 Found RTP audio format 8
 Found RTP audio format 0
 Found RTP audio format 3
 Found RTP audio format 101
 Peer audio RTP is at port 213.166.5.129:10790
 Found description format PCMA
 Found description format PCMU
 Found description format GSM
 Found description format telephone-event
 Capabilities: us - 0x50e(GSM|ULAW|ALAW|G729A|ILBC), peer - audio=0xe(GSM|ULAW|ALAW)/video=0x0(EMPTY), combined - 0xe(GSM|ULAW|ALAW)
 Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723)
 list_route: hop: <sip:07904306012@213.166.5.129>
 set_destination: Parsing <sip:07904306012@213.166.5.129> for address/port to send to
 set_destination: set destination to 213.166.5.129, port 5060
 Transmitting:
ACK sip:07904306012@213.166.5.129 SIP/2.0 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK2fd8e4a4;rport From: <sip:02070433288@incomingnum1.acmechat.org>;tag=as0bdcc570 To: "07904306012" <sip:07904306012@213.166.5.129>;tag=as3d63a8f3 Contact: <sip:02070433288@217.14.132.185> Call-ID: 72bd80002f592b26510108e275c6d18c@213.166.5.129 CSeq: 102 ACK User-Agent: acme VoIP Gateway Content-Length: 0   (NAT) to 213.166.5.129:5060
 
[Ktelth9*CLI>  
Sip read: 
SIP/2.0 200 Ok To: <sip:726002@acmetalk.net>;tag=7ae6ab74 From: "07904306012"<sip:07904306012@217.14.132.162>;tag=as3dbf7d16 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK5adc4dc7 Call-ID: 6182fd661973ee7e02da14820fcfef51@217.14.132.162 CSeq: 103 INVITE Record-Route: <sip:FQSiZLUmOm_BerJMsRtaJS1Mrt7COeB-sRsumHKSydGvWSG@217.14.132.178;ftag=as3dbf7d16;lr=on> Contact: <sip:FQSiZLUmOm_BerJMsRtaJS1Mrt7COeB-sRsumHKSydGvWSG@217.14.132.172:5065;line=ecvb82yn> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, MESSAGE, INFO Content-Type: application/sdp Supported: timer, replaces User-Agent: snom200-3.55 Allow-Events: talk, hold, refer Content-Length: 210  v=0 o=root 138988592 195500907 IN IP4 217.14.132.172 s=call c=IN IP4 217.14.132.172 t=0 0 m=audio 10066 RTP/AVP 8 101 a=fmtp:101 0-15 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv 
 14 headers, 10 lines
 Found RTP audio format 8
 Found RTP audio format 101
 Peer audio RTP is at port 217.14.132.172:10066
 Found description format pcma
 Found description format telephone-event
 Capabilities: us - 0x50e(GSM|ULAW|ALAW|G729A|ILBC), peer - audio=0x8(ALAW)/video=0x0(EMPTY), combined - 0x8(ALAW)
 Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723)
 set_destination: Parsing <sip:726002@217.14.132.178;ftag=as3dbf7d16;lr=on> for address/port to send to
 set_destination: set destination to 217.14.132.178, port 5060
 Transmitting:
ACK sip:FQSiZLUmOm_BerJMsRtaJS1Mrt7COeB-sRsumHKSydGvWSG@217.14.132.172:5065 SIP/2.0 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK7d048bd3 Route: <sip:FQSiZLUmOm_BerJMsRtaJS1Mrt7COeB-sRsumHKSydGvWSG@217.14.132.172:5065;line=ecvb82yn> From: "07904306012" <sip:07904306012@217.14.132.162>;tag=as3dbf7d16 To: <sip:726002@acmetalk.net>;tag=7ae6ab74 Contact: <sip:07904306012@217.14.132.185> Call-ID: 6182fd661973ee7e02da14820fcfef51@217.14.132.162 CSeq: 103 ACK User-Agent: acme VoIP Gateway Content-Length: 0   (no NAT) to 217.14.132.178:5060
 
[Ktelth9*CLI>  
Sip read: 
BYE sip:07904306012@217.14.132.185 SIP/2.0 Record-Route: <sip:07904306012@217.14.132.178;ftag=7ae6ab74;lr=on> To: "07904306012"<sip:07904306012@217.14.132.162>;tag=as3dbf7d16 From: <sip:726002@acmetalk.net>;tag=7ae6ab74 Via: SIP/2.0/UDP 217.14.132.178;branch=z9hG4bK35af.0f558304.0 Via: SIP/2.0/UDP 217.14.132.172:5065;branch=z9hG4bK-d87543-eca67f3575be6a168a82-1-cHBiYjc0OTUxNmE5OTQ2NjZiNTdjYw..-d87543- Call-ID: 6182fd661973ee7e02da14820fcfef51@217.14.132.162 CSeq: 66318031 BYE Contact: <sip:FQSiZLUmOm_BerJMsRtaJS1Mrt7COeB-sRsumHKSydGvWSG@217.14.132.172:5065;line=ecvb82yn> Max-Forwards: 68 User-Agent: snom200-3.55 Content-Length: 0  
 12 headers, 0 lines
 Sending to 217.14.132.178 : 5060 (non-NAT)
 Transmitting (no NAT):
SIP/2.0 200 OK Via: SIP/2.0/UDP 217.14.132.178;branch=z9hG4bK35af.0f558304.0 Via: SIP/2.0/UDP 217.14.132.172:5065;branch=z9hG4bK-d87543-eca67f3575be6a168a82-1-cHBiYjc0OTUxNmE5OTQ2NjZiNTdjYw..-d87543- Record-Route: <sip:07904306012@217.14.132.178;ftag=7ae6ab74;lr=on> From: <sip:726002@acmetalk.net>;tag=7ae6ab74 To: "07904306012"<sip:07904306012@217.14.132.162>;tag=as3dbf7d16 Call-ID: 6182fd661973ee7e02da14820fcfef51@217.14.132.162 CSeq: 66318031 BYE User-Agent: acme VoIP Gateway Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:07904306012@217.14.132.185> Content-Length: 0  
 to 217.14.132.178:5060
 
[Ktelth9*CLI> 
set_destination: Parsing <sip:07904306012@213.166.5.129> for address/port to send to
 set_destination: set destination to 213.166.5.129, port 5060
 We're at 217.14.132.185 port 9816
 Answering with preferred capability 0x8(ALAW)
 Answering with preferred capability 0x4(ULAW)
 Answering with preferred capability 0x400(ILBC)
 Answering with preferred capability 0x2(GSM)
 Answering with preferred capability 0x100(G729A)
 Answering with non-codec capability 0x1(G723)
 11 headers, 14 lines
 Reliably Transmitting:
INVITE sip:07904306012@213.166.5.129 SIP/2.0 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK6c247b09;rport From: <sip:02070433288@incomingnum1.acmechat.org>;tag=as0bdcc570 To: "07904306012" <sip:07904306012@213.166.5.129>;tag=as3d63a8f3 Contact: <sip:02070433288@217.14.132.185> Call-ID: 72bd80002f592b26510108e275c6d18c@213.166.5.129 CSeq: 103 INVITE User-Agent: acme VoIP Gateway Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 318  v=0 o=root 28181 28183 IN IP4 217.14.132.185 s=session c=IN IP4 217.14.132.185 t=0 0 m=audio 9816 RTP/AVP 8 0 97 3 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - -  (NAT) to 213.166.5.129:5060
 
[Ktelth9*CLI> 
  == Spawn extension (prepaid-dial, s, 1) exited non-zero on 'SIP/213.166.5.129-47a6a2a8'
 
[Ktelth9*CLI> 
Feb 18 15:59:12 [31;40mERROR[0;37;40m[28181]: [1;37;40mcdr_addon_mysql.c[0;37;40m:[1;37;40m122[0;37;40m [1;37;40mmysql_log[0;37;40m:  cdr_mysql: cannot connect to database server localhost.  Call will not be logged
 
[Ktelth9*CLI>  
Sip read: 
SIP/2.0 200 OK Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK6c247b09 From: <sip:02070433288@incomingnum1.acmechat.org>;tag=as0bdcc570 To: "07904306012" <sip:07904306012@213.166.5.129>;tag=as3d63a8f3 Call-ID: 72bd80002f592b26510108e275c6d18c@213.166.5.129 CSeq: 103 INVITE User-Agent: MSS VoIP Gateway Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:07904306012@213.166.5.129> Content-Type: application/sdp Content-Length: 265  v=0 o=root 30728 30730 IN IP4 213.166.5.129 s=session c=IN IP4 213.166.5.129 t=0 0 m=audio 10790 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - 
 11 headers, 12 lines
 Found RTP audio format 8
 Found RTP audio format 0
 Found RTP audio format 3
 Found RTP audio format 101
 Peer audio RTP is at port 213.166.5.129:10790
 Found description format PCMA
 Found description format PCMU
 Found description format GSM
 Found description format telephone-event
 Capabilities: us - 0x50e(GSM|ULAW|ALAW|G729A|ILBC), peer - audio=0xe(GSM|ULAW|ALAW)/video=0x0(EMPTY), combined - 0xe(GSM|ULAW|ALAW)
 Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723)
 set_destination: Parsing <sip:07904306012@213.166.5.129> for address/port to send to
 set_destination: set destination to 213.166.5.129, port 5060
 Transmitting:
ACK sip:07904306012@213.166.5.129 SIP/2.0 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK02fcb206;rport 
[Ktelth9*CLI> 
From: <sip:02070433288@incomingnum1.acmechat.org>;tag=as0bdcc570 To: "07904306012" <sip:07904306012@213.166.5.129>;tag=as3d63a8f3 Contact: <sip:02070433288@217.14.132.185> Call-ID: 72bd80002f592b26510108e275c6d18c@213.166.5.129 CSeq: 103 ACK User-Agent: acme VoIP Gateway Content-Length: 0   (NAT) to 213.166.5.129:5060
 set_destination: Parsing <sip:07904306012@213.166.5.129> for address/port to send to
 set_destination: set destination to 213.166.5.129, port 5060
 Reliably Transmitting:
BYE sip:07904306012@213.166.5.129 SIP/2.0 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK455a1e9b;rport From: <sip:02070433288@incomingnum1.acmechat.org>;tag=as0bdcc570 To: "07904306012" <sip:07904306012@213.166.5.129>;tag=as3d63a8f3 Contact: <sip:02070433288@217.14.132.185> Call-ID: 72bd80002f592b26510108e275c6d18c@213.166.5.129 CSeq: 104 BYE User-Agent: acme VoIP Gateway Content-Length: 0   (NAT) to 213.166.5.129:5060
 Destroying call '6182fd661973ee7e02da14820fcfef51@217.14.132.162'
 
[Ktelth9*CLI>  
Sip read: 
SIP/2.0 200 OK Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK455a1e9b From: <sip:02070433288@incomingnum1.acmechat.org>;tag=as0bdcc570 To: "07904306012" <sip:07904306012@213.166.5.129>;tag=as3d63a8f3 Call-ID: 72bd80002f592b26510108e275c6d18c@213.166.5.129 CSeq: 104 BYE User-Agent: MSS VoIP Gateway Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:07904306012@213.166.5.129> Content-Length: 0  
 10 headers, 0 lines
 Destroying call '72bd80002f592b26510108e275c6d18c@213.166.5.129'
 
[Ktelth9*CLI> 
    -- Hungup 'IAX2/83.216.129.33:4569/3'
   == Spawn extension (prepaid-dial, s, 1) exited non-zero on 'SIP/213.166.5.129-08325360'
 
[Ktelth9*CLI> 
Feb 18 15:59:44 [31;40mERROR[0;37;40m[26926]: [1;37;40mcdr_addon_mysql.c[0;37;40m:[1;37;40m122[0;37;40m [1;37;40mmysql_log[0;37;40m:  cdr_mysql: cannot connect to database server localhost.  Call will not be logged
 set_destination: Parsing <sip:02077234465@213.166.5.129> for address/port to send to
 set_destination: set destination to 213.166.5.129, port 5060
 Reliably Transmitting:
BYE sip:02077234465@213.166.5.129 SIP/2.0 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK040ead21;rport From: <sip:02070438202@incomingnum1.acmechat.org>;tag=as0534c91c To: "02077234465" <sip:02077234465@213.166.5.129>;tag=as442c9200 Contact: <sip:02070438202@217.14.132.185> Call-ID: 6b00de9b6cdc26ef56da68f70ab6f9dd@213.166.5.129 CSeq: 102 BYE User-Agent: acme VoIP Gateway Content-Length: 0   (NAT) to 213.166.5.129:5060
 
[Ktelth9*CLI>  
Sip read: 
SIP/2.0 200 OK Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK040ead21 From: <sip:02070438202@incomingnum1.acmechat.org>;tag=as0534c91c To: "02077234465" <sip:02077234465@213.166.5.129>;tag=as442c9200 Call-ID: 6b00de9b6cdc26ef56da68f70ab6f9dd@213.166.5.129 CSeq: 102 BYE User-Agent: MSS VoIP Gateway Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:02077234465@213.166.5.129> Content-Length: 0  
 10 headers, 0 lines
 Message is BYE
 Destroying call '6b00de9b6cdc26ef56da68f70ab6f9dd@213.166.5.129'
 
[Ktelth9*CLI>  
Sip read: 
INVITE sip:02070438290@incomingnum1.acmechat.org SIP/2.0 Via: SIP/2.0/UDP 213.166.5.129:5060;branch=z9hG4bK48ce727e From: "01494720626" <sip:01494720626@213.166.5.129>;tag=as061a6855 To: <sip:02070438290@incomingnum1.acmechat.org> Contact: <sip:01494720626@213.166.5.129> Call-ID: 6c904f0f6cba7d2d4e33456c156ff02c@213.166.5.129 CSeq: 102 INVITE User-Agent: MSS VoIP Gateway Date: Fri, 18 Feb 2005 15:52:56 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 265  v=0 o=root 30766 30766 IN IP4 213.166.5.129 s=session c=IN IP4 213.166.5.129 t=0 0 m=audio 30386 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - 
 12 headers, 12 lines
 Using latest request as basis request
 Sending to 213.166.5.129 : 5060 (NAT)
 Found RTP audio format 8
 Found RTP audio format 0
 Found RTP audio format 3
 Found RTP audio format 101
 Peer audio RTP is at port 213.166.5.129:30386
 Found description format PCMA
 Found description format PCMU
 Found description format GSM
 Found description format telephone-event
 Capabilities: us - 0x50e(GSM|ULAW|ALAW|G729A|ILBC), peer - audio=0xe(GSM|ULAW|ALAW)/video=0x0(EMPTY), combined - 0xe(GSM|ULAW|ALAW)
 Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723)
 Found no matching peer or user for '213.166.5.129:5060'
 Looking for 02070438290 in prepaid
 list_route: hop: <sip:01494720626@213.166.5.129>
 Transmitting (NAT):
SIP/2.0 100 Trying Via: SIP/2.0/UDP 213.166.5.129:5060;branch=z9hG4bK48ce727e;received=213.166.5.129;rport=5060 From: "01494720626" <sip:01494720626@213.166.5.129>;tag=as061a6855 To: <sip:02070438290@incomingnum1.acmechat.org>;tag=as7cc14de3 Call-ID: 6c904f0f6cba7d2d4e33456c156ff02c@213.166.5.129 CSeq: 102 INVITE User-Agent: acme VoIP Gateway Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:02070438290@217.14.132.185> Content-Length: 0  
 to 213.166.5.129:5060
 
[Ktelth9*CLI> 
    -- Executing [1;36;40mAGI[0;37;40m("[1;35;40mSIP/213.166.5.129-47a6a2a8[0;37;40m", "[1;35;40mlookup_dialplan.pl|02070438290[0;37;40m") in new stack
 
[Ktelth9*CLI> 
    -- Launched AGI Script /var/lib/asterisk/agi-bin/lookup_dialplan.pl
 
[Ktelth9*CLI> 
    -- AGI Script lookup_dialplan.pl completed, returning 0
 
[Ktelth9*CLI> 
    -- Executing [1;36;40mDial[0;37;40m("[1;35;40mSIP/213.166.5.129-47a6a2a8[0;37;40m", "[1;35;40mIAX2/02070438290@213.161.73.33/02070438290[0;37;40m") in new stack
 
[Ktelth9*CLI> 
    -- Called 02070438290@213.161.73.33/02070438290
 
[Ktelth9*CLI> 
    -- Call accepted by 213.161.73.33 (format GSM)
     -- Format for call is GSM
 
[Ktelth9*CLI> 
    -- IAX2/213.161.73.33:4569/5 answered SIP/213.166.5.129-47a6a2a8
 
[Ktelth9*CLI> 
We're at 217.14.132.185 port 10330
 
[Ktelth9*CLI> 
Answering with preferred capability 0x8(ALAW)
 
[Ktelth9*CLI> 
Answering with preferred capability 0x4(ULAW)
 
[Ktelth9*CLI> 
Answering with preferred capability 0x400(ILBC)
 
[Ktelth9*CLI> 
Answering with preferred capability 0x2(GSM)
 
[Ktelth9*CLI> 
Answering with preferred capability 0x100(G729A)
 
[Ktelth9*CLI> 
Answering with non-codec capability 0x1(G723)
 
[Ktelth9*CLI> 
Reliably Transmitting (NAT):
SIP/2.0 200 OK Via: SIP/2.0/UDP 213.166.5.129:5060;branch=z9hG4bK48ce727e;received=213.166.5.129;rport=5060 From: "01494720626" <sip:01494720626@213.166.5.129>;tag=as061a6855 To: <sip:02070438290@incomingnum1.acmechat.org>;tag=as7cc14de3 Call-ID: 6c904f0f6cba7d2d4e33456c156ff02c@213.166.5.129 CSeq: 102 INVITE User-Agent: acme VoIP Gateway Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:02070438290@217.14.132.185> Content-Type: application/sdp Content-Length: 319  v=0 o=root 28186 28186 IN IP4 217.14.132.185 s=session c=IN IP4 217.14.132.185 t=0 0 m=audio 10330 RTP/AVP 8 0 97 3 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - 
 to 213.166.5.129:5060
 
[Ktelth9*CLI>  
Sip read: 
ACK sip:02070438290@incomingnum1.acmechat.org SIP/2.0 Via: SIP/2.0/UDP 213.166.5.129:5060;branch=z9hG4bK126ae13d From: "01494720626" <sip:01494720626@213.166.5.129>;tag=as061a6855 To: <sip:02070438290@incomingnum1.acmechat.org>;tag=as7cc14de3 Contact: <sip:01494720626@213.166.5.129> Call-ID: 6c904f0f6cba7d2d4e33456c156ff02c@213.166.5.129 CSeq: 102 ACK User-Agent: MSS VoIP Gateway Content-Length: 0  
 9 headers, 0 lines
 
[Ktelth9*CLI> 
    -- Hungup 'IAX2/83.216.129.33:4569/8'
   == Spawn extension (prepaid-dial, s, 1) exited non-zero on 'SIP/213.166.5.129-47a0bf70'
 
[Ktelth9*CLI> 
Feb 18 16:00:15 [31;40mERROR[0;37;40m[28106]: [1;37;40mcdr_addon_mysql.c[0;37;40m:[1;37;40m122[0;37;40m [1;37;40mmysql_log[0;37;40m:  cdr_mysql: cannot connect to database server localhost.  Call will not be logged
 set_destination: Parsing <sip:01943831293@213.166.5.129> for address/port to send to
 set_destination: set destination to 213.166.5.129, port 5060
 Reliably Transmitting:
BYE sip:01943831293@213.166.5.129 SIP/2.0 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK738c6dfc;rport From: <sip:02070438202@incomingnum1.acmechat.org>;tag=as4138d857 To: "01943831293" <sip:01943831293@213.166.5.129>;tag=as17cc7df6 Contact: <sip:02070438202@217.14.132.185> Call-ID: 0c4682540c4b9eb61ac2a6ad70397072@213.166.5.129 CSeq: 102 BYE User-Agent: acme VoIP Gateway Content-Length: 0   (NAT) to 213.166.5.129:5060
 
[Ktelth9*CLI>  
Sip read: 
SIP/2.0 200 OK Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK738c6dfc From: <sip:02070438202@incomingnum1.acmechat.org>;tag=as4138d857 To: "01943831293" <sip:01943831293@213.166.5.129>;tag=as17cc7df6 Call-ID: 0c4682540c4b9eb61ac2a6ad70397072@213.166.5.129 CSeq: 102 BYE User-Agent: MSS VoIP Gateway Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:01943831293@213.166.5.129> Content-Length: 0  
 10 headers, 0 lines
 Message is BYE
 Destroying call '0c4682540c4b9eb61ac2a6ad70397072@213.166.5.129'
 
[Ktelth9*CLI>  telth9*CLI>  telth9*CLI>  telth9*CLI>  telth9*CLI>  telth9*CLI>  telth9*CLI>  telth9*CLI>  telth9*CLI>  telth9*CLI>  telth9*CLI>  telth9*CLI>  telth9*CLI>  telth9*CLI>  telth9*CLI>  telth9*CLI>  telth9*CLI>  telth9*CLI>  telth9*CLI>  telth9*CLI>  telth9*CLI>  telth9*CLI>  
Sip read: 
INVITE sip:02070433288@incomingnum1.acmechat.org SIP/2.0 Via: SIP/2.0/UDP 213.166.5.129:5060;branch=z9hG4bK2d144752 From: "07904306012" <sip:07904306012@213.166.5.129>;tag=as2f9251bc To: <sip:02070433288@incomingnum1.acmechat.org> Contact: <sip:07904306012@213.166.5.129> Call-ID: 1e2cf93f1d4c57c533be93ca75525292@213.166.5.129 CSeq: 102 INVITE User-Agent: MSS VoIP Gateway Date: Fri, 18 Feb 2005 15:53:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 265  v=0 o=root 30791 30791 IN IP4 213.166.5.129 s=session c=IN IP4 213.166.5.129 t=0 0 m=audio 13204 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - 
 12 headers, 12 lines
 Using latest request as basis request
 Sending to 213.166.5.129 : 5060 (NAT)
 Found RTP audio format 8
 Found RTP audio format 0
 Found RTP audio format 3
 Found RTP audio format 101
 Peer audio RTP is at port 213.166.5.129:13204
 Found description format PCMA
 Found description format PCMU
 Found description format GSM
 Found description format telephone-event
 Capabilities: us - 0x50e(GSM|ULAW|ALAW|G729A|ILBC), peer - audio=0xe(GSM|ULAW|ALAW)/video=0x0(EMPTY), combined - 0xe(GSM|ULAW|ALAW)
 Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723)
 Found no matching peer or user for '213.166.5.129:5060'
 Looking for 02070433288 in prepaid
 list_route: hop: <sip:07904306012@213.166.5.129>
 Transmitting (NAT):
SIP/2.0 100 Trying Via: SIP/2.0/UDP 213.166.5.129:5060;branch=z9hG4bK2d144752;received=213.166.5.129;rport=5060 From: "07904306012" <sip:07904306012@213.166.5.129>;tag=as2f9251bc To: <sip:02070433288@incomingnum1.acmechat.org>;tag=as7b07a9a3 Call-ID: 1e2cf93f1d4c57c533be93ca75525292@213.166.5.129 CSeq: 102 INVITE User-Agent: acme VoIP Gateway Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:02070433288@217.14.132.185> Content-Length: 0  
 to 213.166.5.129:5060
 
[Ktelth9*CLI> 
    -- Executing [1;36;40mAGI[0;37;40m("[1;35;40mSIP/213.166.5.129-47a72100[0;37;40m", "[1;35;40mlookup_dialplan.pl|02070433288[0;37;40m") in new stack
 
[Ktelth9*CLI> 
    -- Launched AGI Script /var/lib/asterisk/agi-bin/lookup_dialplan.pl
 
[Ktelth9*CLI> 
    -- AGI Script lookup_dialplan.pl completed, returning 0
 
[Ktelth9*CLI> 
    -- Executing [1;36;40mDial[0;37;40m("[1;35;40mSIP/213.166.5.129-47a72100[0;37;40m", "[1;35;40mSIP/726002@acmetalk.net[0;37;40m") in new stack
 
[Ktelth9*CLI> 
We're at 217.14.132.185 port 11276
 
[Ktelth9*CLI> 
Answering/Requesting with root capability 8
 
[Ktelth9*CLI> 
Answering with preferred capability 0x4(ULAW)
 
[Ktelth9*CLI> 
Answering with preferred capability 0x400(ILBC)
 
[Ktelth9*CLI> 
Answering with preferred capability 0x2(GSM)
 
[Ktelth9*CLI> 
Answering with preferred capability 0x100(G729A)
 
[Ktelth9*CLI> 
Answering with non-codec capability 0x1(G723)
 
[Ktelth9*CLI> 
12 headers, 14 lines
 
[Ktelth9*CLI> 
Reliably Transmitting:
INVITE sip:726002@acmetalk.net SIP/2.0 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK6a089dde From: "07904306012" <sip:07904306012@217.14.132.162>;tag=as4609f449 To: <sip:726002@acmetalk.net> Contact: <sip:07904306012@217.14.132.185> Call-ID: 16eb40500ff6d0dd4a706a7d5ae7dffe@217.14.132.162 CSeq: 102 INVITE User-Agent: acme VoIP Gateway Date: Fri, 18 Feb 2005 16:00:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 319  v=0 o=root 28195 28195 IN IP4 217.14.132.185 s=session c=IN IP4 217.14.132.185 t=0 0 m=audio 11276 RTP/AVP 8 0 97 3 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - -  (no NAT) to 217.14.132.178:5060
 
[Ktelth9*CLI> 
    -- Called 726002@acmetalk.net
 
[Ktelth9*CLI>  
Sip read: 
SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK6a089dde From: "07904306012" <sip:07904306012@217.14.132.162>;tag=as4609f449 To: <sip:726002@acmetalk.net> Call-ID: 16eb40500ff6d0dd4a706a7d5ae7dffe@217.14.132.162 CSeq: 102 INVITE Server: Sip EXpress router (0.8.12-1rc6 (i386/linux)) Content-Length: 0 Warning: 392 217.14.132.178:5060 "Noisy feedback tells:  pid=17619 req_src_ip=217.14.132.185 req_src_port=5060 in_uri=sip:726002@acmetalk.net out_uri=sip:FIXytGX26r8RqmKcYcuK9f2cfg4S6TCuYcv-SKJCGQ.@217.14.132.172:5065;line=ecvb82yn via_cnt==1"  
 9 headers, 0 lines
 
[Ktelth9*CLI>  
Sip read: 
SIP/2.0 180 Ringing To: <sip:726002@acmetalk.net> From: "07904306012"<sip:07904306012@217.14.132.162>;tag=as4609f449 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK6a089dde Call-ID: 16eb40500ff6d0dd4a706a7d5ae7dffe@217.14.132.162 CSeq: 102 INVITE Record-Route: <sip:726002@217.14.132.178;ftag=as4609f449;lr=on> Contact: <sip:FQSiZLUmOm_BerJMsRtaJS1Mrt7COeB-sRsumHKSydGvWSG@217.14.132.172:5065;line=ecvb82yn> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0  
 11 headers, 0 lines
 
[Ktelth9*CLI> 
    -- SIP/acmetalk.net-99ec is ringing
 
[Ktelth9*CLI> 
Transmitting (NAT):
SIP/2.0 180 Ringing Via: SIP/2.0/UDP 213.166.5.129:5060;branch=z9hG4bK2d144752;received=213.166.5.129;rport=5060 From: "07904306012" <sip:07904306012@213.166.5.129>;tag=as2f9251bc To: <sip:02070433288@incomingnum1.acmechat.org>;tag=as7b07a9a3 Call-ID: 1e2cf93f1d4c57c533be93ca75525292@213.166.5.129 CSeq: 102 INVITE User-Agent: acme VoIP Gateway Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:02070433288@217.14.132.185> Content-Length: 0  
 to 213.166.5.129:5060
 
[Ktelth9*CLI>  
Sip read: 
SIP/2.0 180 Ringing To: <sip:726002@acmetalk.net> From: "07904306012"<sip:07904306012@217.14.132.162>;tag=as4609f449 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK6a089dde Call-ID: 16eb40500ff6d0dd4a706a7d5ae7dffe@217.14.132.162 CSeq: 102 INVITE Record-Route: <sip:726002@217.14.132.178;ftag=as4609f449;lr=on> Contact: <sip:FQSiZLUmOm_BerJMsRtaJS1Mrt7COeB-sRsumHKSydGvWSG@217.14.132.172:5065;line=ecvb82yn> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0  
 11 headers, 0 lines
 
[Ktelth9*CLI> 
    -- SIP/acmetalk.net-99ec is ringing
 
[Ktelth9*CLI>  
Sip read: 
SIP/2.0 180 Ringing To: <sip:726002@acmetalk.net> From: "07904306012"<sip:07904306012@217.14.132.162>;tag=as4609f449 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK6a089dde Call-ID: 16eb40500ff6d0dd4a706a7d5ae7dffe@217.14.132.162 CSeq: 102 INVITE Record-Route: <sip:726002@217.14.132.178;ftag=as4609f449;lr=on> Contact: <sip:FQSiZLUmOm_BerJMsRtaJS1Mrt7COeB-sRsumHKSydGvWSG@217.14.132.172:5065;line=ecvb82yn> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0  
 11 headers, 0 lines
 
[Ktelth9*CLI> 
    -- SIP/acmetalk.net-99ec is ringing
 
[Ktelth9*CLI>  
Sip read: 
SIP/2.0 180 Ringing To: <sip:726002@acmetalk.net> From: "07904306012"<sip:07904306012@217.14.132.162>;tag=as4609f449 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK6a089dde Call-ID: 16eb40500ff6d0dd4a706a7d5ae7dffe@217.14.132.162 CSeq: 102 INVITE Record-Route: <sip:726002@217.14.132.178;ftag=as4609f449;lr=on> Contact: <sip:FQSiZLUmOm_BerJMsRtaJS1Mrt7COeB-sRsumHKSydGvWSG@217.14.132.172:5065;line=ecvb82yn> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0  
 11 headers, 0 lines
 
[Ktelth9*CLI> 
    -- SIP/acmetalk.net-99ec is ringing
 
[Ktelth9*CLI>  
Sip read: 
SIP/2.0 180 Ringing To: <sip:726002@acmetalk.net> From: "07904306012"<sip:07904306012@217.14.132.162>;tag=as4609f449 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK6a089dde Call-ID: 16eb40500ff6d0dd4a706a7d5ae7dffe@217.14.132.162 CSeq: 102 INVITE Record-Route: <sip:726002@217.14.132.178;ftag=as4609f449;lr=on> Contact: <sip:FQSiZLUmOm_BerJMsRtaJS1Mrt7COeB-sRsumHKSydGvWSG@217.14.132.172:5065;line=ecvb82yn> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0  
 11 headers, 0 lines
 
[Ktelth9*CLI> 
    -- SIP/acmetalk.net-99ec is ringing
 
[Ktelth9*CLI>  
Sip read: 
SIP/2.0 200 Ok To: <sip:726002@acmetalk.net>;tag=79dff17c From: "07904306012"<sip:07904306012@217.14.132.162>;tag=as4609f449 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK6a089dde Call-ID: 16eb40500ff6d0dd4a706a7d5ae7dffe@217.14.132.162 CSeq: 102 INVITE Record-Route: <sip:726002@217.14.132.178;ftag=as4609f449;lr=on> Contact: <sip:FQSiZLUmOm_BerJMsRtaJS1Mrt7COeB-sRsumHKSydGvWSG@217.14.132.172:5065;line=ecvb82yn> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, MESSAGE, INFO Content-Type: application/sdp Supported: timer, replaces User-Agent: snom200-3.55 Allow-Events: talk, hold, refer Content-Length: 211  v=0 o=root 1914840843 837029371 IN IP4 217.14.132.172 s=call c=IN IP4 217.14.132.172 t=0 0 m=audio 10094 RTP/AVP 8 101 a=fmtp:101 0-15 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv 
 14 headers, 10 lines
 Found RTP audio format 8
 Found RTP audio format 101
 Peer audio RTP is at port 217.14.132.172:10094
 Found description format pcma
 Found description format telephone-event
 Capabilities: us - 0x50e(GSM|ULAW|ALAW|G729A|ILBC), peer - audio=0x8(ALAW)/video=0x0(EMPTY), combined - 0x8(ALAW)
 Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723)
 list_route: hop: <sip:726002@217.14.132.178;ftag=as4609f449;lr=on>
 list_route: hop: <sip:FQSiZLUmOm_BerJMsRtaJS1Mrt7COeB-sRsumHKSydGvWSG@217.14.132.172:5065;line=ecvb82yn>
 set_destination: Parsing <sip:726002@217.14.132.178;ftag=as4609f449;lr=on> for address/port to send to
 set_destination: set destination to 217.14.132.178, port 5060
 Transmitting:
ACK sip:726002@acmetalk.net SIP/2.0 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK53f0f39e Route: <sip:FQSiZLUmOm_BerJMsRtaJS1Mrt7COeB-sRsumHKSydGvWSG@217.14.132.172:5065;line=ecvb82yn> From: "07904306012" <sip:07904306012@217.14.132.162>;tag=as4609f449 To: <sip:726002@acmetalk.net>;tag=79dff17c Contact: <sip:07904306012@217.14.132.185> Call-ID: 16eb40500ff6d0dd4a706a7d5ae7dffe@217.14.132.162 CSeq: 102 ACK User-Agent: acme VoIP Gateway Content-Length: 0   (no NAT) to 217.14.132.178:5060
 
[Ktelth9*CLI> 
    -- SIP/acmetalk.net-99ec answered SIP/213.166.5.129-47a72100
 We're at 217.14.132.185 port 11646
 Answering with preferred capability 0x8(ALAW)
 Answering with preferred capability 0x4(ULAW)
 Answering with preferred capability 0x400(ILBC)
 Answering with preferred capability 0x2(GSM)
 Answering with preferred capability 0x100(G729A)
 Answering with non-codec capability 0x1(G723)
 Reliably Transmitting (NAT):
SIP/2.0 200 OK Via: SIP/2.0/UDP 213.166.5.129:5060;branch=z9hG4bK2d144752;received=213.166.5.129;rport=5060 From: "07904306012" <sip:07904306012@213.166.5.129>;tag=as2f9251bc To: <sip:02070433288@incomingnum1.acmechat.org>;tag=as7b07a9a3 Call-ID: 1e2cf93f1d4c57c533be93ca75525292@213.166.5.129 CSeq: 102 INVITE User-Agent: acme VoIP Gateway Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:02070433288@217.14.132.185> Content-Type: application/sdp Content-Length: 319  v=0 o=root 28195 28195 IN IP4 217.14.132.185 s=session c=IN IP4 217.14.132.185 t=0 0 m=audio 11646 RTP/AVP 8 0 97 3 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - 
 to 213.166.5.129:5060
     -- Attempting native bridge of SIP/213.166.5.129-47a72100 and SIP/acmetalk.net-99ec
 set_destination: Parsing <sip:726002@217.14.132.178;ftag=as4609f449;lr=on> for address/port to send to
 set_destination: set destination to 217.14.132.178, port 5060
 We're at 217.14.132.185 port 11276
 Answering/Requesting with root capability 8
 Answering with preferred capability 0x4(ULAW)
 Answering with preferred capability 0x2(GSM)
 Answering with non-codec capability 0x1(G723)
 12 headers, 12 lines
 Reliably Transmitting:
INVITE sip:FQSiZLUmOm_BerJMsRtaJS1Mrt7COeB-sRsumHKSydGvWSG@217.14.132.172:5065 SIP/2.0 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK68855379 Route: <sip:FQSiZLUmOm_BerJMsRtaJS1Mrt7COeB-sRsumHKSydGvWSG@217.14.132.172:5065;line=ecvb82yn> From: "07904306012" <sip:07904306012@217.14.132.162>;tag=as4609f449 To: <sip:726002@acmetalk.net>;tag=79dff17c Contact: <sip:07904306012@217.14.132.185> Call-ID: 16eb40500ff6d0dd4a706a7d5ae7dffe@217.14.132.162 CSeq: 103 INVITE User-Agent: acme VoIP Gateway Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 265  v=0 o=root 28195 28196 IN IP4 213.166.5.129 s=session c=IN IP4 213.166.5.129 t=0 0 m=audio 13204 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - -  (no NAT) to 217.14.132.178:5060
 
[Ktelth9*CLI>  
Sip read: 
SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK68855379 From: "07904306012" <sip:07904306012@217.14.132.162>;tag=as4609f449 To: <sip:726002@acmetalk.net>;tag=79dff17c Call-ID: 16eb40500ff6d0dd4a706a7d5ae7dffe@217.14.132.162 CSeq: 103 INVITE Server: Sip EXpress router (0.8.12-1rc6 (i386/linux)) Content-Length: 0  
 8 headers, 0 lines
 
[Ktelth9*CLI>  
Sip read: 
ACK sip:02070433288@incomingnum1.acmechat.org SIP/2.0 Via: SIP/2.0/UDP 213.166.5.129:5060;branch=z9hG4bK78b52e7a From: "07904306012" <sip:07904306012@213.166.5.129>;tag=as2f9251bc To: <sip:02070433288@incomingnum1.acmechat.org>;tag=as7b07a9a3 Contact: <sip:07904306012@213.166.5.129> Call-ID: 1e2cf93f1d4c57c533be93ca75525292@213.166.5.129 CSeq: 102 ACK User-Agent: MSS VoIP Gateway Content-Length: 0  
 9 headers, 0 lines
 set_destination: Parsing <sip:07904306012@213.166.5.129> for address/port to send to
 set_destination: set destination to 213.166.5.129, port 5060
 We're at 217.14.132.185 port 11646
 Answering with preferred capability 0x8(ALAW)
 Answering with non-codec capability 0x1(G723)
 11 headers, 10 lines
 Reliably Transmitting:
INVITE sip:07904306012@213.166.5.129 SIP/2.0 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK07a35c28;rport From: <sip:02070433288@incomingnum1.acmechat.org>;tag=as7b07a9a3 To: "07904306012" <sip:07904306012@213.166.5.129>;tag=as2f9251bc Contact: <sip:02070433288@217.14.132.185> Call-ID: 1e2cf93f1d4c57c533be93ca75525292@213.166.5.129 CSeq: 102 INVITE User-Agent: acme VoIP Gateway Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 220  v=0 o=root 28195 28196 IN IP4 217.14.132.172 s=session c=IN IP4 217.14.132.172 t=0 0 m=audio 10094 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - -  (NAT) to 213.166.5.129:5060
 
[Ktelth9*CLI>  
Sip read: 
SIP/2.0 200 OK Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK07a35c28 From: <sip:02070433288@incomingnum1.acmechat.org>;tag=as7b07a9a3 To: "07904306012" <sip:07904306012@213.166.5.129>;tag=as2f9251bc Call-ID: 1e2cf93f1d4c57c533be93ca75525292@213.166.5.129 CSeq: 102 INVITE User-Agent: MSS VoIP Gateway Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:07904306012@213.166.5.129> Content-Type: application/sdp Content-Length: 265  v=0 o=root 30791 30792 IN IP4 213.166.5.129 s=session c=IN IP4 213.166.5.129 t=0 0 m=audio 13204 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - 
 11 headers, 12 lines
 Found RTP audio format 8
 Found RTP audio format 0
 Found RTP audio format 3
 Found RTP audio format 101
 Peer audio RTP is at port 213.166.5.129:13204
 Found description format PCMA
 Found description format PCMU
 Found description format GSM
 Found description format telephone-event
 Capabilities: us - 0x50e(GSM|ULAW|ALAW|G729A|ILBC), peer - audio=0xe(GSM|ULAW|ALAW)/video=0x0(EMPTY), combined - 0xe(GSM|ULAW|ALAW)
 Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723)
 list_route: hop: <sip:07904306012@213.166.5.129>
 set_destination: Parsing <sip:07904306012@213.166.5.129> for address/port to send to
 set_destination: set destination to 213.166.5.129, port 5060
 Transmitting:
ACK sip:07904306012@213.166.5.129 SIP/2.0 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK2d7faa9b;rport From: <sip:02070433288@incomingnum1.acmechat.org>;tag=as7b07a9a3 To: "07904306012" <sip:07904306012@213.166.5.129>;tag=as2f9251bc Contact: <sip:02070433288@217.14.132.185> Call-ID: 1e2cf93f1d4c57c533be93ca75525292@213.166.5.129 CSeq: 102 ACK User-Agent: acme VoIP Gateway Content-Length: 0   (NAT) to 213.166.5.129:5060
 
[Ktelth9*CLI>  
Sip read: 
SIP/2.0 491 Request Pending To: <sip:726002@acmetalk.net>;tag=79dff17c From: "07904306012"<sip:07904306012@217.14.132.162>;tag=as4609f449 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK68855379 Call-ID: 16eb40500ff6d0dd4a706a7d5ae7dffe@217.14.132.162 CSeq: 103 INVITE Max-Forwards: 9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER User-Agent: acme VoIP Gateway Content-Length: 0  
 10 headers, 0 lines
     -- Got SIP response 491 "Request Pending" back from 217.14.132.178

[Ktelth9*CLI> 
 set_destination: Parsing <sip:726002@217.14.132.178;ftag=as4609f449;lr=on> for address/port to send to
 set_destination: set destination to 217.14.132.178, port 5060
 Transmitting:
ACK sip:FQSiZLUmOm_BerJMsRtaJS1Mrt7COeB-sRsumHKSydGvWSG@217.14.132.172:5065 SIP/2.0 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK68855379 Route: <sip:FQSiZLUmOm_BerJMsRtaJS1Mrt7COeB-sRsumHKSydGvWSG@217.14.132.172:5065;line=ecvb82yn> From: "07904306012" <sip:07904306012@217.14.132.162>;tag=as4609f449 To: <sip:726002@acmetalk.net>;tag=79dff17c Contact: <sip:07904306012@217.14.132.185> Call-ID: 16eb40500ff6d0dd4a706a7d5ae7dffe@217.14.132.162 CSeq: 103 ACK User-Agent: acme VoIP Gateway Content-Length: 0   (no NAT) to 217.14.132.178:5060
 
[Ktelth9*CLI> 
set_destination: Parsing <sip:07904306012@213.166.5.129> for address/port to send to
 set_destination: set destination to 213.166.5.129, port 5060
 We're at 217.14.132.185 port 11646
 Answering with preferred capability 0x8(ALAW)
 Answering with preferred capability 0x4(ULAW)
 Answering with preferred capability 0x400(ILBC)
 Answering with preferred capability 0x2(GSM)
 Answering with preferred capability 0x100(G729A)
 Answering with non-codec capability 0x1(G723)
 11 headers, 14 lines
 Reliably Transmitting:
INVITE sip:07904306012@213.166.5.129 SIP/2.0 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK2982e798;rport From: <sip:02070433288@incomingnum1.acmechat.org>;tag=as7b07a9a3 To: "07904306012" <sip:07904306012@213.166.5.129>;tag=as2f9251bc Contact: <sip:02070433288@217.14.132.185> Call-ID: 1e2cf93f1d4c57c533be93ca75525292@213.166.5.129 CSeq: 103 INVITE User-Agent: acme VoIP Gateway Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 319  v=0 o=root 28195 28197 IN IP4 217.14.132.185 s=session c=IN IP4 217.14.132.185 t=0 0 m=audio 11646 RTP/AVP 8 0 97 3 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - -  (NAT) to 213.166.5.129:5060
 
[Ktelth9*CLI> 
11 headers, 0 lines
 Reliably Transmitting:
REGISTER sip:iax3.hitchhiker.net SIP/2.0 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK70954d2d From: <sip:acme@iax3.hitchhiker.net>;tag=as5d9b5c60 To: <sip:acme@iax3.hitchhiker.net> Call-ID: 44a08a3f44d83edf7ebecfb2138abe69@217.14.132.162 CSeq: 114 REGISTER User-Agent: acme VoIP Gateway Expires: 120 Contact: <sip:s@217.14.132.185> Event: registration Content-Length: 0   (no NAT) to 213.166.5.130:5060
 Destroying call '16eb40500ff6d0dd4a706a7d5ae7dffe@217.14.132.162'
 
[Ktelth9*CLI> 
  == Spawn extension (prepaid-dial, s, 1) exited non-zero on 'SIP/213.166.5.129-47a72100'
 
[Ktelth9*CLI> 
Feb 18 16:00:38 [31;40mERROR[0;37;40m[28195]: [1;37;40mcdr_addon_mysql.c[0;37;40m:[1;37;40m122[0;37;40m [1;37;40mmysql_log[0;37;40m:  cdr_mysql: cannot connect to database server localhost.  Call will not be logged
 
[Ktelth9*CLI>  
Sip read: 
SIP/2.0 100 Trying Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK70954d2d From: <sip:acme@iax3.hitchhiker.net>;tag=as5d9b5c60 To: <sip:acme@iax3.hitchhiker.net>;tag=as7d7e155e Call-ID: 44a08a3f44d83edf7ebecfb2138abe69@217.14.132.162 CSeq: 114 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:acme@213.166.5.130> Content-Length: 0  
 10 headers, 0 lines
 
[Ktelth9*CLI>  
Sip read: 
SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK70954d2d From: <sip:acme@iax3.hitchhiker.net>;tag=as5d9b5c60 To: <sip:acme@iax3.hitchhiker.net>;tag=as7d7e155e Call-ID: 44a08a3f44d83edf7ebecfb2138abe69@217.14.132.162 CSeq: 114 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:acme@213.166.5.130> WWW-Authenticate: Digest realm="asterisk", nonce="49476b92" Content-Length: 0  
 11 headers, 0 lines
 12 headers, 0 lines
 Reliably Transmitting:
REGISTER sip:iax3.hitchhiker.net SIP/2.0 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK6667033a From: <sip:acme@iax3.hitchhiker.net>;tag=as50696a89 To: <sip:acme@iax3.hitchhiker.net> Call-ID: 44a08a3f44d83edf7ebecfb2138abe69@217.14.132.162 CSeq: 115 REGISTER User-Agent: acme VoIP Gateway Authorization: Digest username="acme", realm="asterisk", algorithm=MD5, uri="sip:acme@213.166.5.130", nonce="49476b92", response="d30613ab4aa989472f5f2c971d4e2b11", opaque="" Expires: 120 Contact: <sip:s@217.14.132.185> Event: registration Content-Length: 0   (no NAT) to 213.166.5.130:5060
 
[Ktelth9*CLI>  
Sip read: 
SIP/2.0 200 OK Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK2982e798 From: <sip:02070433288@incomingnum1.acmechat.org>;tag=as7b07a9a3 To: "07904306012" <sip:07904306012@213.166.5.129>;tag=as2f9251bc Call-ID: 1e2cf93f1d4c57c533be93ca75525292@213.166.5.129 CSeq: 103 INVITE User-Agent: MSS VoIP Gateway Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:07904306012@213.166.5.129> Content-Type: application/sdp Content-Length: 265  v=0 o=root 30791 30793 IN IP4 213.166.5.129 s=session c=IN IP4 213.166.5.129 t=0 0 m=audio 13204 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - 
 11 headers, 12 lines
 Found RTP audio format 8
 Found RTP audio format 0
 Found RTP audio format 3
 Found RTP audio format 101
 Peer audio RTP is at port 213.166.5.129:13204
 Found description format PCMA
 Found description format PCMU
 Found description format GSM
 Found description format telephone-event
 Capabilities: us - 0x50e(GSM|ULAW|ALAW|G729A|ILBC), peer - audio=0xe(GSM|ULAW|ALAW)/video=0x0(EMPTY), combined - 0xe(GSM|ULAW|ALAW)
 Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723)
 set_destination: Parsing <sip:07904306012@213.166.5.129> for address/port to send to
 set_destination: set destination to 213.166.5.129, port 5060
 Transmitting:
ACK sip:07904306012@213.166.5.129 SIP/2.0 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK5d509380;rport 
[Ktelth9*CLI> 
From: <sip:02070433288@incomingnum1.acmechat.org>;tag=as7b07a9a3 To: "07904306012" <sip:07904306012@213.166.5.129>;tag=as2f9251bc Contact: <sip:02070433288@217.14.132.185> Call-ID: 1e2cf93f1d4c57c533be93ca75525292@213.166.5.129 CSeq: 103 ACK User-Agent: acme VoIP Gateway Content-Length: 0   (NAT) to 213.166.5.129:5060
 set_destination: Parsing <sip:07904306012@213.166.5.129> for address/port to send to
 set_destination: set destination to 213.166.5.129, port 5060
 Reliably Transmitting:
BYE sip:07904306012@213.166.5.129 SIP/2.0 Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK162187ad;rport From: <sip:02070433288@incomingnum1.acmechat.org>;tag=as7b07a9a3 To: "07904306012" <sip:07904306012@213.166.5.129>;tag=as2f9251bc Contact: <sip:02070433288@217.14.132.185> Call-ID: 1e2cf93f1d4c57c533be93ca75525292@213.166.5.129 CSeq: 104 BYE User-Agent: acme VoIP Gateway Content-Length: 0   (NAT) to 213.166.5.129:5060
 
[Ktelth9*CLI>  
Sip read: 
SIP/2.0 100 Trying Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK6667033a From: <sip:acme@iax3.hitchhiker.net>;tag=as50696a89 To: <sip:acme@iax3.hitchhiker.net>;tag=as7d7e155e Call-ID: 44a08a3f44d83edf7ebecfb2138abe69@217.14.132.162 CSeq: 115 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:acme@213.166.5.130> Content-Length: 0  
 10 headers, 0 lines
 
[Ktelth9*CLI>  
Sip read: 
SIP/2.0 200 OK Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK162187ad From: <sip:02070433288@incomingnum1.acmechat.org>;tag=as7b07a9a3 To: "07904306012" <sip:07904306012@213.166.5.129>;tag=as2f9251bc Call-ID: 1e2cf93f1d4c57c533be93ca75525292@213.166.5.129 CSeq: 104 BYE User-Agent: MSS VoIP Gateway Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:07904306012@213.166.5.129> Content-Length: 0  
 10 headers, 0 lines
 Destroying call '1e2cf93f1d4c57c533be93ca75525292@213.166.5.129'
 
[Ktelth9*CLI>  
Sip read: 
SIP/2.0 200 OK Via: SIP/2.0/UDP 217.14.132.185:5060;branch=z9hG4bK6667033a From: <sip:acme@iax3.hitchhiker.net>;tag=as50696a89 To: <sip:acme@iax3.hitchhiker.net>;tag=as7d7e155e Call-ID: 44a08a3f44d83edf7ebecfb2138abe69@217.14.132.162 CSeq: 115 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 120 Contact: <sip:s@217.14.132.185>;expires=120 Date: Fri, 18 Feb 2005 15:53:46 GMT Content-Length: 0  
 12 headers, 0 lines
 Destroying call '44a08a3f44d83edf7ebecfb2138abe69@217.14.132.162'
 
[Ktelth9*CLI>  
Sip read: 
BYE sip:07904306012@217.14.132.185 SIP/2.0 Record-Route: <sip:07904306012@217.14.132.178;ftag=79dff17c;lr=on> To: "07904306012"<sip:07904306012@217.14.132.162>;tag=as4609f449 From: <sip:726002@acmetalk.net>;tag=79dff17c Via: SIP/2.0/UDP 217.14.132.178;branch=z9hG4bK0a91.6561eb61.0 Via: SIP/2.0/UDP 217.14.132.172:5065;branch=z9hG4bK-d87543-f9a05f4cc388a84e24db-1-cHAwYTQ2OGYzZjg5NTZmZTI2NjQ4Nw..-d87543- Call-ID: 16eb40500ff6d0dd4a706a7d5ae7dffe@217.14.132.162 CSeq: 60260540 BYE Contact: <sip:FQSiZLUmOm_BerJMsRtaJS1Mrt7COeB-sRsumHKSydGvWSG@217.14.132.172:5065;line=ecvb82yn> Max-Forwards: 68 User-Agent: snom200-3.55 Content-Length: 0  
 12 headers, 0 lines
 Sending to 217.14.132.178 : 5060 (NAT)
 Transmitting (NAT):
SIP/2.0 200 OK Via: SIP/2.0/UDP 217.14.132.178;branch=z9hG4bK0a91.6561eb61.0;received=217.14.132.178;rport=5060 Via: SIP/2.0/UDP 217.14.132.172:5065;branch=z9hG4bK-d87543-f9a05f4cc388a84e24db-1-cHAwYTQ2OGYzZjg5NTZmZTI2NjQ4Nw..-d87543- Record-Route: <sip:07904306012@217.14.132.178;ftag=79dff17c;lr=on> From: <sip:726002@acmetalk.net>;tag=79dff17c To: "07904306012"<sip:07904306012@217.14.132.162>;tag=as4609f449 Call-ID: 16eb40500ff6d0dd4a706a7d5ae7dffe@217.14.132.162 CSeq: 60260540 BYE User-Agent: acme VoIP Gateway Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact:  Content-Length: 0  
 to 217.14.132.178:5060
 Destroying call '16eb40500ff6d0dd4a706a7d5ae7dffe@217.14.132.162'
 
[Ktelth9*CLI> quit ]0;root@telth9:/tmp[root@telth9 tmp]# 
]0;root@telth9:/tmp[root@telth9 tmp]# 
]0;root@telth9:/tmp[root@telth9 tmp]# 
]0;root@telth9:/tmp[root@telth9 tmp]# 
]0;root@telth9:/tmp[root@telth9 tmp]# 
Script done on Fri 18 Feb 2005 16:00:47 GMT